Network Working Group S. Proust
Internet-Draft Orange
Intended status: Informational December 2, 2015
Expires: June 4, 2016
Additional WebRTC audio codecs for interoperability.
draft-ietf-rtcweb-audio-codecs-for-interop-03
Abstract
To ensure a baseline level of interoperability between WebRTC
clients, a minimum set of required codecs is specified. However, to
maximize the possibility to establish 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 clients to address the most relevant
interoperability use cases.
Status of This Memo
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This Internet-Draft will expire on June 4, 2016.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Definition and abbreviations . . . . . . . . . . . . . . . . 3
3. Rationale for additional WebRTC codecs . . . . . . . . . . . 3
4. Additional suitable codecs for WebRTC . . . . . . . . . . . . 5
4.1. AMR-WB . . . . . . . . . . . . . . . . . . . . . . . . . 5
4.1.1. AMR-WB General description . . . . . . . . . . . . . 5
4.1.2. WebRTC relevant use case for AMR-WB . . . . . . . . . 5
4.1.3. Guidelines for AMR-WB usage and implementation with
WebRTC . . . . . . . . . . . . . . . . . . . . . . . 5
4.2. AMR . . . . . . . . . . . . . . . . . . . . . . . . . . . 6
4.2.1. AMR General description . . . . . . . . . . . . . . . 6
4.2.2. WebRTC relevant use case for AMR . . . . . . . . . . 6
4.2.3. Guidelines for AMR usage and implementation with
WebRTC . . . . . . . . . . . . . . . . . . . . . . . 7
4.3. G.722 . . . . . . . . . . . . . . . . . . . . . . . . . . 7
4.3.1. G.722 General description . . . . . . . . . . . . . . 7
4.3.2. WebRTC relevant use case for G.722 . . . . . . . . . 7
4.3.3. Guidelines for G.722 usage and implementation . . . . 8
4.4. Other codecs . . . . . . . . . . . . . . . . . . . . . . 8
5. Security Considerations . . . . . . . . . . . . . . . . . . . 8
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 8
8. References . . . . . . . . . . . . . . . . . . . . . . . . . 8
8.1. Normative references . . . . . . . . . . . . . . . . . . 9
8.2. Informative references . . . . . . . . . . . . . . . . . 10
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 11
1. Introduction
As indicated in [I-D.ietf-rtcweb-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 to establish the session without the need
for audio transcoding, it is recommended in [I-D.ietf-rtcweb-audio]
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 clients to address
the most relevant interoperability use cases.
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The codecs considered in this document are recommended to be
supported and included in the Offer only for WebRTC clients for which
interoperability with other non-WebRTC endpoints and non-WebRTC based
services is relevant as described in Section 4.1.2, Section 4.2.2,
Section 4.3.2. Other use cases may justify offering other additional
codecs to avoid transcodings.
2. Definition 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 radio access (VoLTE) [IR.92].
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 Subsystem
o LTE: Long Term Evolution (3GPP "4G" wireless data transmission
standard)
o MOS: Mean Opinion Score
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 clients can
guarantee codec interoperability (without transcoding) at state of
the art voice quality (better than narrow band "PSTN" quality)
between WebRTC clients. The WebRTC technology is also expected to be
used to communicate with other types of clients 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
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[TS23.002] or DECT based VoIP telephony [EN300175-1]. Consequently,
a significant number of calls are likely to occur between terminals
supporting WebRTC clients and other terminals like mobile handsets,
fixed VoIP terminals, DECT terminals that do not support WebRTC
clients 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 drawbacks of
such transcoding operations are recalled 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 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 fall back 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 less than desirable or
acceptable choice for customers. If transcoding is performed
between OPUS and any other wideband codec, wideband communication
could be maintained but with degraded quality (MOS scores of
transcoding between AMR-WB 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 audio artifacts due to
packet losses, and the audio effects resulting from 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 re-sending. The delays produced
by all these operations are additive and may increase the end to
end delay beyond acceptable limits like with more than 1s end to
end latency.
o Additional costs in networks: transcoding places important
additional costs on network gateways mainly related to codec
implementation, codecs license, deployments, testing and
validation costs. It must be noted that transcoding of wideband
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to wideband would require more CPU processing and be more costly
than between narrowband codecs.
4. Additional suitable codecs for WebRTC
The following codecs are considered as relevant suitable codecs with
respect to the general purpose described in Section 3. This list
reflects the current status of WebRTC foreseen use cases. It is not
limitative and opened to further inclusion of other codecs for which
relevant use cases can be identified. These additional codecs are
recommended to be included in the offer 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 like 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
detailed codec description and Source code are in [TS26.171],
[TS26.173], [TS26.190], [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 implemented in several hundreds of devices
models and 145 HD mobile networks in 85 countries with a customer
base of more than 450 millions. A high number of calls are
consequently likely to occur between WebRTC clients and mobile 3GPP
terminals. The use of AMR-WB by WebRTC clients would consequently
allow transcoding free interoperation with all mobile 3GPP wideband
terminals. Besides, WebRTC clients running on mobile terminals
(smartphones) may reuse the AMR-WB codec already implemented on these
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 bandwidth efficient format and one speech frame encapsulated in
each RTP packets. 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/
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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
client offering AMR-WB it is important that the WebRTC client:
o Offer AMR in addition to AMR-WB with AMR-WB, being a wideband
codec, listed first as preferred payload type with respect to
other narrow band codecs (AMR, G.711...)and with 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. Offer at least one
AMR-WB configuration with parameter settings as defined in
Table 6.1 of [TS26.114]. In order to maximize the
interoperability and quality this offer does not restrict the
codec modes offered. Restrictions in 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, i.e. several hundred millions of terminals. This
include both mobile phone calls using GSM and 3G cellular systems as
well as multimedia telephony services over IP/IMS and 4G/VoLTE, such
as GSMA voice IMS profile for VoLTE in [IR.92]. In addition to
impacts listed above, support of AMR can avoid degrading the high
efficiency over mobile radio access.References for AMR related
specifications including detailed codec description and Source code
are in [TS26.071], [TS26.073], [TS26.090], [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 is
still a high number of legacy terminals supporting only AMR
(terminals with no wideband / HD Voice capabilities) are still used.
The use of AMR by WebRTC client would consequently allow transcoding
free interoperation with all mobile 3GPP terminals. Besides, WebRTC
client running on mobile terminals (smartphones) may reuse the AMR
codec already implemented on these devices.
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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 packets. 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
client offering AMR, it is important that the WebRTC client:
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 Table 6.1 and Table 6.2 of [TS26.114]. In order to
maximize the interoperability and quality this offer shall not
restrict AMR codec modes offered. Restrictions in 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 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.
Especially, G.722 has been chosen by ETSI DECT as the mandatory
wideband codec for New Generation DECT with purpose to greatly
increase the voice quality by extending the bandwidth from narrow
band to wideband. G.722 is the wideband codec required for CAT-iq
DECT certified terminals and the V2.0 of CAT-iq specifications have
been approved by GSMA as minimum requirements for 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. The
market for DECT cordless phones including DECT chipset is more than
150 Millions per year and CAT-IQ is a registered trade make in 47
countries worldwide. G.722 has also been specified by ETSI in
[TS181005] as mandatory wideband codec for IMS multimedia telephony
communication service and supplementary services using fixed
broadband access. The support of G.722 would consequently allow
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transcoding free IP interoperation between WebRTC client and fixed
VoIP terminals including DECT / CAT-IQ terminals supporting G.722.
Besides, WebRTC client running on fixed terminals implementing G.722
may reuse the G.722 codec already implemented on these devices.
4.3.3. Guidelines for G.722 usage and implementation
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 G.722 is 16 kHz
but the rtp clock rate is set to 8000Hz in SDP to stay backward
compatible with an erroneous definition in the original version of
the RTP A/V profile. Further guidelines for implementing and using
G.722 with purpose to ensure interoperability with Multimedia
Telephony services overs IMS can be found in section 7 of [TS26.114].
Additional information of G.722 implementation in DECT can be found
in [EN300175-8] and full codec description and C source code in
[G.722].
4.4. Other codecs
Other interoperability use cases may justify the use of other codecs.
5. Security Considerations
Security considerations for WebRTC Audio Codec and Processing
Requirements can be found in [I-D.ietf-rtcweb-audio]. Implementors
making use of the additional codecs considered in this document are
advised to also report more specifically to the "Security
Considerations" sections of [RFC4867] (for AMR and AMR-WB) and
[RFC3551].
6. IANA Considerations
None.
7. Acknowledgements
Special thanks to Espen Berger, Bernhard Feiten, Bo Burman, Kalyani
Bogineni, Miao Lei, Enrico Marocco, who co-authored the initial
document. Thanks, as well, to Magnus Westerlund and Barry Dingle who
carefully reviewed the document and helped to improve it.
8. References
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8.1. Normative references
[G.722] ITU, "Recommendation ITU-T G.722 (2012): 7 kHz audio-
coding within 64 kbit/s", 2012-09.
[I-D.ietf-rtcweb-audio]
Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
Requirements", draft-ietf-rtcweb-audio-09 (work in
progress), November 2015.
[IR.92] GSMA, "IMS Profile for Voice and SMS V9.0", April 2015.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
DOI 10.17487/RFC3551, July 2003,
<http://www.rfc-editor.org/info/rfc3551>.
[RFC4867] Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q. Xie,
"RTP Payload Format and File Storage Format for the
Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband
(AMR-WB) Audio Codecs", RFC 4867, DOI 10.17487/RFC4867,
April 2007, <http://www.rfc-editor.org/info/rfc4867>.
[TS26.071]
3GPP, "3GPP TS 26.071 v12.0.0: Recommendation ITU-T G.722
(2012): "Mandatory Speech Codec speech processing
functions; AMR Speech CODEC; General description".",
2014-09.
[TS26.073]
3GPP, "3GPP TS 26.073 v12.0.0: ANSI C code for the
Adaptive Multi Rate (AMR) speech codec", 2014-09.
[TS26.090]
3GPP, "3GPP TS 26.090 v12.0.0: Mandatory Speech Codec
speech processing functions; Adaptive Multi-Rate (AMR)
speech codec; Transcoding functions.", 2014-09.
[TS26.104]
3GPP, "3GPP TS 26.104 v12.0.0: ANSI C code for the
floating-point Adaptive Multi Rate (AMR) speech codec.",
2014-09.
[TS26.114]
3GPP, "IP Multimedia Subsystem (IMS); Multimedia
telephony; Media handling and interaction V13.0.0", June
2015.
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[TS26.171]
3GPP, "3GPP TS 26.071 v12.0.0: Recommendation ITU-T G.722
(2012): "Speech codec speech processing functions;
Adaptive Multi-Rate - Wideband (AMR-WB) speech codec;
General description".", 2014-09.
[TS26.173]
3GPP, "3GPP TS 26.073 v12.1.0: ANSI-C code for the
Adaptive Multi-Rate - Wideband (AMR-WB) speech codec.",
2015-03.
[TS26.190]
3GPP, "3GPP TS 26.090 v12.0.0: Speech codec speech
processing functions; Adaptive Multi-Rate - Wideband (AMR-
WB) speech codec; Transcoding functions.", 2014-09.
[TS26.204]
3GPP, "3GPP TS 26.104 v12.1.0: Speech codec speech
processing functions; Adaptive Multi-Rate - Wideband (AMR-
WB) speech codec; ANSI-C code.", 2015-03.
8.2. Informative references
[EN300175-1]
ETSI, "ETSI EN 300 175-1, Digital Enhanced Cordless
Telecommunications (DECT); Common Interface (CI); Part 1:
Overview v2.5.1", 2009.
[EN300175-8]
ETSI, "ETSI EN 300 175-8, v2.5.1: Digital Enhanced
Cordless Telecommunications (DECT); Common Interface (CI);
Part 8: Speech and audio coding and transmission.", 2009.
[G.711] ITU, "Recommendation ITU-T G.711 (2012): Pulse code
modulation (PCM) of voice frequencies", 1988-11.
[I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for
Browser-based Applications", draft-ietf-rtcweb-overview-14
(work in progress), June 2015.
[IR.36] GSMA, "Adaptive Multirate Wide Band V3.0", September 2014.
[RFC6716] Valin, JM., Vos, K., and T. Terriberry, "Definition of the
Opus Audio Codec", RFC 6716, DOI 10.17487/RFC6716,
September 2012, <http://www.rfc-editor.org/info/rfc6716>.
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[RFC7478] Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
Time Communication Use Cases and Requirements", RFC 7478,
DOI 10.17487/RFC7478, March 2015,
<http://www.rfc-editor.org/info/rfc7478>.
[TS181005]
ETSI, "Telecommunications and Internet converged Services
and Protocols for Advanced Networking (TISPAN); Service
and Capability Requirements V3.3.1 (2009-12)", 2009.
[TS23.002]
3GPP, "3GPP TS 23.002 v13.3.0: Network architecture",
2015-09.
Author's Address
Stephane Proust
Orange
2, avenue Pierre Marzin
Lannion 22307
France
Email: stephane.proust@orange.com
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