Network Working Group S. Proust
Internet-Draft Orange
Intended status: Informational E. Berger
Expires: July 20, 2015 Cisco
B. Feiten
Deutsche Telekom
B. Burman
Ericsson
K. Bogineni
Verizon Wireless
M. Lei
Huawei
E. Marocco
Telecom Italia
January 16, 2015
Additional WebRTC audio codecs for interoperability.
draft-ietf-rtcweb-audio-codecs-for-interop-01
Abstract
To ensure a baseline level of interoperability between WebRTC
clients, [I-D.ietf-rtcweb-audio] requires a minimum set of codecs.
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
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
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material or to cite them other than as "work in progress."
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This Internet-Draft will expire on July 20, 2015.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Rationale for additional WebRTC codecs . . . . . . . . . . . 3
4. Additional suitable codecs for WebRTC . . . . . . . . . . . . 4
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 . . . . . . . . . . . . . . . . . . . . . . . 6
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 . . . . 7
4.4. Other codecs . . . . . . . . . . . . . . . . . . . . . . 7
5. Security Considerations . . . . . . . . . . . . . . . . . . . 8
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 8
8. References . . . . . . . . . . . . . . . . . . . . . . . . . 8
8.1. Normative references . . . . . . . . . . . . . . . . . . 8
8.2. Informative references . . . . . . . . . . . . . . . . . 8
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 9
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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.
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 end points and non WebRTC
based services is relevant as described in sections 5.1.2, 5.2.2 and
5.3.2. Other use cases may justify offering other additional codecs
to avoid transcodings. It is the intent of this document to
inventory and document any other additional interoperability use
cases and codecs if needed.
2. Definitions
Legacy networks: In this draft, legacy networks encompass the
conversational networks that are already deployed like the PSTN, the
PLMN, the IMS, H.323 networks.
3. Rationale for additional WebRTC codecs
The mandatory implementation of OPUS [RFC6716] in WebRTC clients can
guarantee the codec interoperability (without transcoding) at the
state of the art voice quality (better than narrow band "PSTN"
quality) between WebRTC clients. The WebRTC technology is however
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 3GPP IMS services (e.g. VoLTE, ViLTE) or to
interoperate with fixed or mobile Circuit Switched or VoIP services
like mobile 3GPP 3G/2G Circuit Switched voice or DECT based VoIP
telephony. 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 at both ends or affected by transcoding operations. The
drawbacks of such transcoding operations are recalled below:
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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 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
12.65kbit/s and OPUS at 16 kbit/s in both directions are
significantly lower than those of AMR-WB at 12.65kbit/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
to wideband would require more CPU 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 4. 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
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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 and 4G/VoLTE, specified by GSMA as voice IMS
profile for VoLTE in [IR.92]. More detailed information on AMR-WB
can be found in [IR.36]. [IR.36] includes references for all 3GPP
AMR-WB related specifications including detailed codec description
and Source code.
4.1.2. WebRTC relevant use case for AMR-WB
The market of voice personal communication is driven by mobile
terminals. AMR-WB is now implemented in more than 200 devices models
and 85 HD mobile networks in 60 countries with a customer base of
more than 100 million. 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
terminal. 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
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
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
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AMR-WB configuration with parameter settings as defined in
Table 6.1 of [TS 26.114]. 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.
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.
4.2.3. Guidelines for AMR usage and implementation with WebRTC
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
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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 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 end-points 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 terminal 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 cordeless 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
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
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.
Some further update of this Draft may provide under this section
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additional use case description and codec implementation guidelines
for these codecs.
5. Security Considerations
6. IANA Considerations
None.
7. Acknowledgements
Thanks to Milan Patel for his review.
8. References
8.1. Normative references
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
8.2. Informative references
[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.722] ITU, "Recommendation ITU-T G.722 (2012): "7 kHz audio-
coding within 64 kbit/s".", 2012.
[I-D.ietf-rtcweb-audio]
Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
Requirements", draft-ietf-rtcweb-audio-07 (work in
progress), October 2014.
[I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for
Browser-based Applications", draft-ietf-rtcweb-overview-13
(work in progress), November 2014.
[I-D.ietf-rtcweb-use-cases-and-requirements]
Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
Time Communication Use-cases and Requirements", draft-
ietf-rtcweb-use-cases-and-requirements-15 (work in
progress), December 2014.
[IR.36] GSMA, "Adaptive Multirate Wide Band", 2013.
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[IR.92] GSMA, "IMS Profile for Voice and SMS", 2013.
[RFC6716] Valin, JM., Vos, K., and T. Terriberry, "Definition of the
Opus Audio Codec", RFC 6716, September 2012.
[TS181005]
ETSI, "Telecommunications and Internet converged Services
and Protocols for Advanced Networking (TISPAN); Service
and Capability Requirements V3.3.1 (2009-12)", 2009.
[TS26.114]
3GPP, "IP Multimedia Subsystem (IMS); Multimedia
telephony; Media handling and interaction", 2011.
Authors' Addresses
Stephane Proust
Orange
2, avenue Pierre Marzin
Lannion 22307
France
Email: stephane.proust@orange.com
Espen Berger
Cisco
Email: espeberg@cisco.com
Bernhard Feiten
Deutsche Telekom
Email: Bernhard.Feiten@telekom.de
Bo Burman
Ericsson
Email: bo.burman@ericsson.com
Kalyani Bogineni
Verizon Wireless
Email: Kalyani.Bogineni@VerizonWireless.com
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Miao Lei
Huawei
Email: lei.miao@huawei.com
Enrico Marocco
Telecom Italia
Email: enrico.marocco@telecomitalia.it
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