Internet DRAFT - draft-marjou-rtcweb-audio-codecs-for-interop
draft-marjou-rtcweb-audio-codecs-for-interop
Network Working Group X. Marjou
Internet-Draft S. Proust
Intended status: Informational France Telecom Orange
Expires: August 29, 2013 K. Bogineni
Verizon Wireless
R. Jesske
B. Feiten
Deutsche Telekom AG
L. Miao
Huawei
E. Enrico
Telecom Italia
E. Berger
Cisco
February 25, 2013
WebRTC audio codecs for interoperability with legacy networks.
draft-marjou-rtcweb-audio-codecs-for-interop-01
Abstract
This document presents use-cases underlining why WebRTC needs AMR-WB,
AMR and G.722 as additional relevant voice codecs to satisfactorily
ensure interoperability with existing systems. It also presents a
way forward that takes into consideration the concerns expressed
against the addition of codecs besides Opus and G.711.
It is especially recognized that unjustified additional costs on
browsers must be avoided. Therefore, the proposed solution intends
to fully rely on the codecs already supported on the devices
implementing the browsers and for which license and implementation
costs have been already paid.
It is expected that this way forward will significantly limit the
costs and technical impacts on browsers while greatly improving
interoperability with legacy systems and overall quality. It intents
to be considered as a good compromise beneficial to all parties and
to the whole industry: the user quality experience will be optimized
as a whole at limited additional costs without incurring high costs
for both networks to support transcoding and browsers to support
additional codecs.
Status of this Memo
This Internet-Draft is submitted to IETF in full conformance with the
provisions of BCP 78 and BCP 79.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 5
3. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 5
4. Use cases . . . . . . . . . . . . . . . . . . . . . . . . . . 5
4.1. AMR-WB . . . . . . . . . . . . . . . . . . . . . . . . . . 5
4.1.1. Use case . . . . . . . . . . . . . . . . . . . . . . . 5
4.1.2. Problem . . . . . . . . . . . . . . . . . . . . . . . 5
4.1.3. Concerns from the browser manufacturers . . . . . . . 7
4.2. AMR . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
4.2.1. Use case . . . . . . . . . . . . . . . . . . . . . . . 7
4.2.2. Problem . . . . . . . . . . . . . . . . . . . . . . . 7
4.2.3. Concerns from the browser manufacturers . . . . . . . 8
4.3. G.722 . . . . . . . . . . . . . . . . . . . . . . . . . . 8
4.3.1. Use case . . . . . . . . . . . . . . . . . . . . . . . 8
4.3.2. Problem . . . . . . . . . . . . . . . . . . . . . . . 8
4.3.3. Concerns from the browser manufacturers . . . . . . . 9
5. The proposed way-forward . . . . . . . . . . . . . . . . . . . 9
6. Security Considerations . . . . . . . . . . . . . . . . . . . 11
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 11
8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 11
9. References . . . . . . . . . . . . . . . . . . . . . . . . . . 11
9.1. Normative references . . . . . . . . . . . . . . . . . . . 11
9.2. Informative references . . . . . . . . . . . . . . . . . . 12
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 12
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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.
In order to reach the agreement to select OPUS and G.711 as mandatory
to implement codecs it was also agreed to consider possible
additional codecs in order to take into account the concerns
expressed on interoperability issues. The discussion is consequently
currently taking place regarding the additional voice/audio codecs
that need to be supported. It is mainly questioned whether Opus and
G.711 are sufficient to properly address the interoperability issues
with legacy systems or if additional codecs need to be supported.
This document presents some use cases highlighting that Opus and
G.711 are not sufficient to properly cover all interoperability
requirements. In Section 4, important interoperability use-cases are
presented describing the interoperability issues that would be
encountered if only Opus and G.711 were supported. It therefore
advocates for the addition of other codecs while addressing concerns
raised against such support.
In section 5, a way forward is proposed that intends to be a real
compromise taking into consideration the concerns expressed against
additional codecs. It is especially recognized that unjustified
additional costs on browsers must be avoided. Therefore the proposed
solution intends to strongly limit the cost and technical impact on
browsers for the support of additional codecs (including license
costs) while improving interoperability with legacy systems.
Regarding audio codecs, it is a common misconception that other
existing voice networks only support G.711. Actually existing
networks use circuit switched networks as well as voice-over-IP
networks like H.323 and SIP-based networks, which means that audio
codecs are not limited to G.711. For instance, from use cases
described in [I-D.ietf-rtcweb-se-ucases-and-requirements], it can be
foreseen that interoperability with mobile telephony systems will
often happen. In such mobile systems, the UE must support the
Adaptive Multi-Rate (AMR) speech codec, and if wideband speech
communication is offered, the UE must support AMR wideband (AMR-WB)
codec. An increasing number of customers are now experiencing high
quality voice with HD Voice services over mobile, fixed networks or
over the internet. For those customers, any fall back to the G.711
narrow band quality for interoperability purpose could be perceived
as a strong and unacceptable quality degradation. Support of G.711
as the only codec for legacy interoperability purposes is currently
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not sufficient.
2. Terminology
In this document, the key words "MUST", "MUST NOT", "REQUIRED",
"SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY",
and "OPTIONAL" are to be interpreted as described in RFC 2119
[RFC2119].
3. 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.
4. Use cases
4.1. AMR-WB
4.1.1. Use case
The market of voice personal communication is driven by mobile
terminals and WebRTC technology is expected to be increasingly used
on smartphones. Furthermore "HD voice" is gaining momentum and more
and more personal communication devices will support wideband
communications. Customers are now getting used to the high quality
offered by HD Voice mobile devices, CAT-iq fixed HD devices and
eventually, HD Voice via WebRTC and OPUS over the internet.
Hence, many communications are expected to be held between a user of
a WebRTC endpoint on a device integrating an AMR-WB module who wants
to communicate with another user that can only be reached on a mobile
device that only supports AMR-WB (and AMR); both endpoints support HD
voice quality.
4.1.2. Problem
For this use case, the best situation will be to have the AMR-WB
supported by both sides of the connection. Indeed, as mentioned in
the introduction, AMR-WB is specified by 3GPP as the mandatory codec
to be supported by wideband mobile terminals for a wide range of
communication services as described in [AMR-WB]. This includes the
massively deployed circuit switched mobile telephony services and new
multimedia telephony services over IP/IMS and 4G/VoLTE as specified
by GSMA as voice IMS profile for VoLTE in IR92. Hence, AMR-WB is
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strongly increasing with deployment in more than 60 networks from 45
countries and more than 130 types of terminals (c.f.
[Information-Papers]).
In that use case, if OPUS and G.711 remain the only codecs supported
by the WebRTC endpoints, a gateway must then transcode these codecs
into AMR-WB, and vice-versa, in order to implement the use-case. As
a consequence, a high number of calls are likely to be affected by
transcoding operations producing a degradation of the user quality
experience for many customers. This will have a very significant
business impact for all service providers on both sides, not only
with respect to the transcoding costs but mainly with respect to user
experience degradation.
The drawbacks of transcoding are recalled below:
o Cost issues: transcoding places important additional costs on
network gateways for example codec implementation and license
costs, deployments costs, testing/validation costs etc... However
these costs can be seen as just transferred from the terminal side
to the network side. The real issue is rather the degradation of
the quality of service affecting the end user perceived quality
which will be harmful to all concerned service providers.
o Intrinsic quality degradation: Subjective test results show that
intrinsic voice quality is significantly degraded by transcoding.
The degradation is around 0.2 to 0.3 MOS for most of transcoding
use cases with AMR-WB at 12.65 kbit/s. It should be stressed that
if transcoding is performed between AMR-WB and G.711, wideband
voice quality will be lost. Such bandwidth reduction effect
clearly degrades the user perceived quality of service leading to
shorter and less frequent calls (see ref_gsma). Such a switch to
G.711 will not be accepted anymore by customers. If transcoding
is performed between AMR-WB and OPUS, wideband communication could
be maintained. However, as the WB codecs complexity is higher
than NB codecs complexity, such WB transcoding is also more costly
and degrades the 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 interactivity due to increased latency: Transcoding means
full de-packetization for decoding of the media stream (including
mechanisms of de-jitter buffering and packet loss recovery) then
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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.
In addition to these drawbacks related to transcoding, the following
issue must be considered:
o Efficiency over mobile radio access: AMR-WB has been designed and
extensively tested for optimized and robust usage over mobile
radio access which results in enhanced capacity and efficiency.
The mobile radio bearer is optimized for such a codec with channel
coding protecting its most sensitive bits. Furthermore, AMR-WB is
more efficient than OPUS at regular bit rates used for mobile
communication of 12.65 kbit/s with fall back modes down to 6.6
kbit/s. Finally, hardware optimized implementation may allow for
less battery consumption
As a consequence, re-using AMR-WB would be beneficial for the
specific usage of WebRTC technology over mobile networks. With the
strong increase of the smartphone market the capability to use such a
mobile codec could strongly enforce and extend the market penetration
of the Web RTC technology.
4.1.3. Concerns from the browser manufacturers
The browser manufacturers are concerned by the additional costs that
the implementation of AMR-WB would put on browsers which include
integration and test costs and codec license costs. The proposed way
forward in Section 5 takes carefully into account this concern.
4.2. AMR
4.2.1. Use case
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 a high number
of legacy terminals supporting only AMR (terminals with no wideband /
HD Voice capabilities) are still used.
4.2.2. Problem
For this use case, the best solution will be to have the AMR
supported by both sides of the connection. Indeed, AMR is specified
by 3GPP as the mandatory codec to be supported by any mobile terminal
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for a wide range of communication services. This includes the
massively deployed circuit switched mobile telephony services and new
multimedia telephony services over IP/IMS and 4G/VoLTE as specified
by GSMA as voice IMS profile for VoLTE in IR92. Hundreds of millions
of terminals are consequently currently supporting AMR and are not
supporting OPUS nor G.711.
In that use case, if OPUS and G.711 remain the only codecs supported
by the WebRTC endpoints the same problem as described in 4.1.1 will
be experienced because of transcoding impacts (costs, quality
degradation and increased latency) and lower efficiency over mobile
radio access.
As a consequence, re-using AMR would be beneficial for the specific
usage of WebRTC technology over mobile networks. With the strong
increase of the smartphone market the capability to use such a mobile
codec could strongly enforce and extend the market penetration of the
Web RTC technology.
4.2.3. Concerns from the browser manufacturers
Same as in Section 4.1.3.
4.3. G.722
4.3.1. Use case
As mentioned in Section 4.1.1, HD Voice is gaining momentum and more
and more personal communication devices support wideband
communications. Customers get used to high quality voice and WebRTC
aims at providing high voice quality over internet. In this use
case, a user of a WebRTC endpoint on a device integrating G.722
module wants to communicate with another user that can only be
reached on a device that only supports G.722 as a wideband codec,
G.722 being specified by ETSI as the mandatory wideband codec for New
Generation DECT (e.g. CAT-iq compliant).
4.3.2. Problem
For this use case, the best solution will be to have the G.722
supported by both sides of the connection.
Indeed, G.722 has been chosen by ETSI DECT to greatly increase the
voice quality by extending the bandwidth from narrow band to
wideband. Besides providing high wideband quality, it has low
complexity and very low delay. It is widely used in HD fixed
services in both hard and soft endpoints.
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In this use case, if OPUS and G.711 remain the only codecs supported
by the WebRTC endpoints, a gateway must then transcode them from and
into G.722 in order to implement the use-case. As in Section 4.1.2,
it should be stressed that if transcoding is performed between G.722
and G.711, wideband quality will be lost with fall back to narrow
band. This will be perceived as a strong and unacceptable quality
degradation by customers experiencing more and more wideband voice
calls. It is also important to recall that wideband audio can help
persons with hearing impairments to use voice communication over
distance and drafts regulations dealing with this requires wide band
audio wherever there is voice communication pointing at G.722 as the
common codec at least to assure interoperability with wide-band audio
between providers.
On the other hand, transcoding with OPUS will greatly increase the
complexity, now, as mentioned above, G.722 low complexity was a key
factor in many applications mandating G.722.
4.3.3. Concerns from the browser manufacturers
Unlike AMR and AMR-WB, G.722, as G.711, is royalty free. Some
concerns about the availability of G.722 PLC were raised. Indeed
G.722 and G.711 were initially designed without Packet Loss
Concealment (PLC); nevertheless, ITU-T did standardize such
functionality as appendices to these recommendations to extend the
capabilities of current systems to support new applications and to
follow the market demand (for instance when these standards have been
widely used in VoIP applications). It has been argued that, unlike
the main recommendations, there are non-RF IPR declarations for these
PLC appendices (G.711 Appendix I, G.722 Appendices III and IV). It
should be recalled that these appendices are only examples and
implementers are free to use any PLC solution. For instance in the
G.722 case, there are publicly available PLC in the ITU-T Software
Tool Library.
5. The proposed way-forward
It is proposed that the browser manufacturers re-use AMR, AMR-WB and
G.722 codecs whenever they are already supported on the device on
which the browsers are implemented.
AMR and AMR-WB are already supported by millions of devices with
license costs and technical costs (implementation, tests...) already
paid for these codecs optimized for mobile usage.
Android now provides the APIs needed to give access to all the voice
and audio features implemented on the hardware that are required to
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develop a voice service. This especially includes AMR and AMR-WB
encoding and decoding, adaptive gain control, echo cancellation and
noise reduction.
It is therefore technically feasible that browsers offer AMR and
AMR-WB as additional RTC Web codecs without re-implementing these
codecs and so with very limited additional costs:
o For implementation, no codec software code has to be developed,
tested or integrated directly in the browser (neither for the
encoding/decoding nor for related audio functions).
o For licensing, fees are already paid for the hardware
implementation. Since no additional codec is implemented on the
device with only one active voice call using the codec at a time
it cannot be argued that additional license fees have to be paid
in that case. However, in order to fully guarantee this, it is
made explicit in the proposed text that the support of AMR and
AMR-WB is conditional on the fact that no additional license fee
is required.
This proposed way forward is expected to be a good compromise
beneficial to all parties. It will optimize the user experience with
limited additional costs: excluding high costs for networks to
support transcoding and high additional costs on browsers to support
additional codecs.
It is consequently proposed the following wording to be added to
Section 3 (codec requirements) of [I-D.ietf-rtcweb-audio-codecs]:
3. Audio Codecs
3.1. Required Codecs
To ensure a baseline level of interoperability between WebRTC
clients, a minimum set of required codecs are specified below.
WebRTC clients are REQUIRED to implement the following audio
codecs.
* Opus [RFC6716], with any ptime value up to 120 ms
* G.711 PCMA and PCMU with one channel, a rate of 8000 Hz and a
ptime of 20 - see section 4.5.14 of [RFC3551]
* Telephone Event - [RFC4733]
3.2. Additional Codecs
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To ensure an enhanced level of interoperability between WebRTC
clients, AMR-WB, AMR and G.722 codecs SHOULD be implemented by
WebRTC end-points to avoid transcoding costs and quality
degradations towards legacy fixed and mobile devices and allow
interworking with enhanced voice quality (rather than fall back to
G.711 narrow band voice).
WebRTC browsers on devices for which the implementation of AMR is
mandatory for voice services MUST allow AMR to be negotiated and
used at WebRTC level provided it is ensured that no additional
license fees are required.
WebRTC browsers on wide-band devices for which the implementation
of AMR-WB is mandatory for voice services MUST allow AMR-WB to be
negotiated and used at WebRTC level provided it is ensured that no
additional license fees are required.
WebRTC browsers devices for which the implementation of G.722 is
mandatory for voice services MUST allow G.722 to be negotiated and
used at WebRTC level.
Note: the wording of section "3.2. Additional Codecs" is a first
proposal and example to try to capture the general principle
explained in this document to improve interoperability while limiting
the cost impact on browsers. It is subject to further modifications
to reach the best possible compromise.
6. Security Considerations
7. IANA Considerations
None.
8. Acknowledgements
Thanks to Milan Patel for his review.
9. References
9.1. Normative references
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
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[RFC2616] Fielding, R., Gettys, J., Mogul, J., Frystyk, H.,
Masinter, L., Leach, P., and T. Berners-Lee, "Hypertext
Transfer Protocol -- HTTP/1.1", RFC 2616, June 1999.
9.2. Informative references
[AMR-WB] GSMA, "AMR-WB", 2011.
[I-D.ietf-rtcweb-audio]
Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
Requirements", draft-ietf-rtcweb-audio-01 (work in
progress), November 2012.
[I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for Brower-
based Applications", draft-ietf-rtcweb-overview-06 (work
in progress), February 2013.
[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-10 (work in
progress), December 2012.
[Information-Papers]
GSMA, "Information Papers", 2013.
Authors' Addresses
Xavier Marjou
France Telecom Orange
2, avenue Pierre Marzin
Lannion 22307
France
Email: xavier.marjou@orange.com
Stephane Proust
France Telecom Orange
2, avenue Pierre Marzin
Lannion 22307
France
Email: stephane.proust@orange.com
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Kalyani Bogineni
Verizon Wireless
Email: Kalyani.Bogineni@VerizonWireless.com
Roland Jesske
Deutsche Telekom AG
Email: R.Jesske@telekom.de
Bernhard Feiten
Deutsche Telekom AG
Email: R.Jesske@telekom.de
Lei Miao
Huawei
Email: lei.miao@huawei.com
Marocco
Telecom Italia
Email: enrico.marocco@telecomitalia.it
Espen Berger
Cisco
Email: espeberg@cisco.com
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