wdiff rfc7875.original rfc7875.txt

Network Working Group
Internet Engineering Task Force (IETF) S. Proust, Ed.
Internet-Draft
Request for Comments: 7875 Orange
Intended status:
Category: Informational April 22, 2016
Expires: October 24, May 2016
ISSN: 2070-1721

Additional WebRTC audio codecs Audio Codecs for interoperability.
draft-ietf-rtcweb-audio-codecs-for-interop-06 Interoperability

Abstract

To ensure a baseline level of interoperability between WebRTC endpoints, a
minimum set of required codecs is specified. However, to maximize
the possibility to establish 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
interoperability use cases. to interoperability.

Status of This Memo

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provisions of BCP 78 and BCP 79.

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This Internet-Draft will expire on October 24, 2016.
http://www.rfc-editor.org/info/rfc7875.

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Table of Contents

1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Definition Definitions and abbreviations Abbreviations . . . . . . . . . . . . . . . . 3
3. Rationale for additional Additional WebRTC codecs Codecs . . . . . . . . . . . 3
4. Additional suitable codecs Suitable Codecs for WebRTC . . . . . . . . . . . . 5
4.1. AMR-WB . . . . . . . . . . . . . . . . . . . . . . . . . 5
4.1.1. AMR-WB General description Description . . . . . . . . . . . . . 5
4.1.2. WebRTC relevant use case WebRTC-Relevant Use Case for AMR-WB . . . . . . . . . 5
4.1.3. Guidelines for AMR-WB usage Usage and implementation Implementation with
WebRTC . . . . . . . . . . . . . . . . . . . . . . . 5
4.2. AMR . . . . . . . . . . . . . . . . . . . . . . . . . . . 6
4.2.1. AMR General description Description . . . . . . . . . . . . . . . 6
4.2.2. WebRTC relevant use case WebRTC-Relevant Use Case for AMR . . . . . . . . . . 6
4.2.3. Guidelines for AMR usage Usage and implementation Implementation with
WebRTC . . . . . . . . . . . . . . . . . . . . . . . 6
4.3. G.722 . . . . . . . . . . . . . . . . . . . . . . . . . . 7
4.3.1. G.722 General description Description . . . . . . . . . . . . . . 7
4.3.2. WebRTC relevant use case WebRTC-Relevant Use Case for G.722 . . . . . . . . . 7
4.3.3. Guidelines for G.722 usage Usage and implementation Implementation . . . . 8
5. Security Considerations . . . . . . . . . . . . . . . . . . . 8
6. IANA Considerations References . . . . . . . . . . . . . . . . . . . . . 8
7. Acknowledgements . . . . 8
6.1. Normative References . . . . . . . . . . . . . . . . . . 8
8.
6.2. Informative References . . . . . . . . . . . . . . . . . 10
Acknowledgements . . . . . . . . . . . 9
8.1. Normative references . . . . . . . . . . . . . 11
Contributors . . . . . 9
8.2. Informative references . . . . . . . . . . . . . . . . . 10 . . . . 11
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 11

1. Introduction

As indicated in [I-D.ietf-rtcweb-overview], [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 of establishing the session without the need for audio
transcoding, it is recommended in [I-D.ietf-rtcweb-audio] [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 interoperability use cases. to interoperability.

The codecs considered in this document are recommended to be
supported and included in the Offer offer, only for WebRTC endpoints for
which interoperability with other non-WebRTC endpoints and non-WebRTC
based non-
WebRTC-based services is relevant as described in Section Sections 4.1.2,
Section
4.2.2, Section and 4.3.2. Other use cases may justify offering other
additional codecs to avoid transcoding.

2. Definition Definitions and abbreviations 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 (VoLTE) [IR.92].

o WebRTC endpoint: a A WebRTC endpoint can be a WebRTC browser or a
WebRTC non browser non-browser (also called "WebRTC device" or "WebRTC native
application") as defined in [I-D.ietf-rtcweb-overview] [OVERVIEW].

o AMR: Adaptive Multi-Rate. Multi-Rate

o AMR-WB: Adaptive Multi-Rate WideBand. Wideband

o CAT-iq: Cordless Advanced Technology-internet Technology - internet and quality. quality

o DECT: Digital Enhanced Cordless Telecommunications

o IMS: IP Multimedia Subsystem Subsystems

o LTE: Long Term Evolution (3GPP "4G" wireless data transmission
standard)

o MOS: Mean Opinion Score, defined in ITU-T Recommendation P.800 specification
[P.800]

o PSTN: Public Switched Telephone Network

o PLMN: Public Land Mobile Network

o VoLTE: Voice Over over LTE

3. Rationale for additional Additional WebRTC codecs Codecs

The mandatory implementation of OPUS Opus [RFC6716] in WebRTC endpoints
can guarantee codec interoperability (without transcoding) at state
of the art state-
of-the-art voice quality (better than narrow band 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 Circuit-Switched or VoIP services
like mobile Circuit Switched Circuit-Switched voice over 3GPP 2G/3G mobile networks
[TS23.002] or DECT based 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 terminals, and DECT terminals that do not support WebRTC
endpoints nor implement OPUS. Opus. As a consequence, these calls are
likely to be either of low narrow band 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 fall back fallback codec
for interoperation, wideband voice quality will be lost. Such
bandwidth reduction effect down to narrow band clearly degrades
the user perceived user-perceived quality of service leading to shorter and less
frequent calls. Such a switch to G.711 is less than desirable or
acceptable a choice for customers.
If transcoding is performed between OPUS Opus and any other wideband
codec, wideband communication could be maintained but with
degraded quality (MOS scores (MOSs of transcoding between AMR-WB at 12.65
kbit/s and OPUS Opus at 16 kbit/s in both directions are significantly
lower than those of AMR-WB at 12.65 kbit/s or OPUS Opus at 16 kbit/s).
Furthermore, in degraded conditions, the addition of defects, like
(a) audio artifacts due to packet losses, losses and the (b) audio effects resulting from due
to the cascading of different packet loss recovery algorithms 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 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 re-packetization, and re-sending. resending. The delays produced
by all these operations are additive and may increase the end to 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 Suitable Codecs for WebRTC

The following codecs are considered as relevant codecs with respect to the
general purpose described in Section 3. This list reflects the
current status of WebRTC foreseen use cases. cases for WebRTC. It is not
limitative and opened
limitative; it is open to further inclusion of other codecs for which
relevant use cases can be identified. These additional codecs are It's recommended to be included in the offer in include
codecs (in addition to OPUS Opus and G.711 G.711) according to the foreseen
interoperability cases to be addressed.

4.1. AMR-WB

4.1.1. AMR-WB General description Description

The Adaptive Multi-Rate WideBand (AMR-WB) is a 3GPP defined 3GPP-defined speech
codec that is mandatory to implement in any 3GPP terminal that
supports wideband speech communication. It is being used in circuit 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 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 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" A voice", where "HD" stands for "High
Definition". Consequently, a high number of calls are consequently likely to
occur between WebRTC endpoints and mobile 3GPP terminals offering
AMR-WB. The Thus, the use of AMR-WB by WebRTC endpoints would
consequently allow transcoding free
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 these those
devices.

4.1.3. Guidelines for AMR-WB usage Usage and implementation Implementation with WebRTC

The payload format to be used for AMR-WB is described in [RFC4867]
with bandwidth efficient a bandwidth-efficient format and one speech frame encapsulated
in each RTP packet. Further guidelines for implementing and using AMR-
WB
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 AMR-WB, it is important that the WebRTC endpoints:

o Offer AMR in addition to AMR-WB AMR-WB, with AMR-WB listed first (AMR-WB
being a wideband codec) as the preferred payload type with respect
to other narrow band narrow-band codecs (AMR, G.711...) G.711, etc.) and with Bandwidth
Efficient a
bandwidth-efficient payload format preferred.

o Be capable of operating AMR-WB with any subset of the nine codec
modes and source controlled 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 the
interoperability and quality quality, this offer does not restrict the
codec modes offered. Restrictions in on the use of codec modes may
be included in the answer.

4.2. AMR

4.2.1. AMR General description Description

Adaptive Multi-Rate (AMR) is a 3GPP defined 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]. In addition to the impacts listed above, support of AMR can
avoid degrading the high efficiency over mobile radio access.
References for AMR related AMR-related specifications including detailed codec
description and source code are in [TS26.071], [TS26.073], [TS26.090],
and [TS26.104].

4.2.2. WebRTC relevant use case 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 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 Usage and implementation 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 source-controlled rate operation.

o Offer at least one configuration with parameter settings as
defined in Table Tables 6.1 and Table 6.2 of [TS26.114]. In order to maximize
the interoperability and quality quality, this offer shall not restrict
AMR codec modes offered. Restrictions in on the use of codec modes
may be included in the answer.

4.3. G.722

4.3.1. G.722 General description Description

G.722 [G.722] is an ITU-T defined ITU-T-defined wideband speech codec. G.722 was
approved by the ITU-T in 1988. It is a royalty free 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] [EN300175-8], which is 2.5 to 3 times lower than
AMR-WB.
Especially, In particular, G.722 has been chosen by ETSI DECT as the
mandatory wideband codec for New Generation DECT with purpose 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 CAT-iq
DECT certified terminals
that are certified as CAT-iq DECT, and the V2.0 version 2.0 of the CAT-iq
specifications have been approved by GSMA as the minimum requirements
for HD voice the "HD voice" logo usage on "fixed" devices; devices, i.e., broadband
connections using the G.722 codec.

4.3.2. WebRTC relevant use case WebRTC-Relevant Use Case for G.722

G.722 is the wideband codec required for DECT CAT-iq terminals. DECT
cordeless
cordless phones are still widely used to offer short range 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 transcoding-free IP interoperation between WebRTC
endpoints and fixed VoIP terminals including DECT / CAT-IQ CAT-iq terminals
supporting G.722. Besides, WebRTC endpoints running on fixed
terminals
implementing that implement G.722 may reuse the G.722 codec already
implemented on these devices.

4.3.3. Guidelines for G.722 usage Usage and implementation 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 the G.722 codec
is 16 kHz kHz, but the rtp RTP clock rate is set to 8000Hz 8000 Hz in SDP to stay
backward compatible with an erroneous definition in the original
version of the RTP A/V audio/video profile. Further guidelines for
implementing and using G.722 with purpose to ensure interoperability with
multimedia telephony services over IMS can be found in section Section 7 of
[TS26.114]. Additional information of about the G.722 implementation in
DECT can be found in [EN300175-8] [EN300175-8], and the full codec description and
C source code are in [G.722].

5. Security Considerations

Security

Relevant security considerations for WebRTC Audio Codec and Processing
Requirements can be found in [I-D.ietf-rtcweb-audio]. Implementors [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].

6. IANA Considerations

None.

7. Acknowledgements

The authors of this document are

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.

The authors would like to thank Magnus Westerlund, Barry Dingle and
Sanjay Mishra who carefully reviewed [RFC3551] (for the document and helped to
improve it.

8. RTP audio/video
profile).

6. References

8.1.

6.1. Normative references References

[G.722] ITU, "Recommendation ITU-T G.722 (2012): 7 ITU-T, "7 kHz audio-
coding audio-coding within 64 kbit/s", 2012-09, ITU-T
Recommendation G.722, September 2012,
<http://www.itu.int/rec/T-REC-G.722-201209-I/en>.

[I-D.ietf-rtcweb-audio]
Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
Requirements", draft-ietf-rtcweb-audio-10 (work in
progress), February 2016.

[IR.92] GSMA, "IMS Profile for Voice and SMS V9.0", SMS", IR.92, Version 9.0,
April 2015, <http://www.gsma.com/newsroom/all-documents/
ir-92-ims-profile-for-voice-and-sms/>.

[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>.

[RFC7874] Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
Requirements", RFC 7874, DOI 10.17487/RFC7874, May 2016,
<http://www.rfc-editor.org/info/rfc7874>.

[TS26.071]
3GPP, "3GPP TS 26.071 v13.0.0: Recommendation ITU-T G.722
(2012): "Mandatory Speech Codec speech processing functions;
AMR Speech CODEC; General description".",
2015-12, description", 3GPP TS 26.171
v13.0.0, December 2015,
<http://www.3gpp.org/DynaReport/26071.htm>.

[TS26.073]
3GPP, "3GPP TS 26.073 v13.0.0: ANSI "ANSI C code for the Adaptive Multi Rate (AMR)
speech codec", 2015-12, 3GPP TS 26.073 v13.0.0, December 2015,
<http://www.3gpp.org/DynaReport/26073.htm>.

[TS26.090]
3GPP, "3GPP TS 26.090 v13.0.0: Mandatory "Mandatory Speech Codec speech processing functions;
Adaptive Multi-Rate (AMR) speech codec; Transcoding
functions.", 2015-12, 3GPP TS 26.090 v13.0.0, December 2015,
<http://www.3gpp.org/DynaReport/26090.htm>.

[TS26.104]
3GPP, "3GPP TS 26.104 v13.0.0: ANSI "ANSI C code for the floating-point Adaptive Multi
Rate (AMR) speech codec.",
2015-12, 3GPP TS 26.104 v13.0.0,
December 2015, <http://www.3gpp.org/DynaReport/26090.htm>.

[TS26.114]
3GPP, "3GPP TS 26.114 v13.3.0: IP "IP Multimedia Subsystem (IMS); Multimedia
telephony; Media handling and interaction", 3GPP TS 26.114
v13.3.0, March 2016,
<http://www.3gpp.org/DynaReport/26114.htm>.

[TS26.171]
3GPP, "3GPP TS 26.171 v13.0.0: Speech "Speech codec speech processing functions; Adaptive
Multi-Rate - Wideband (AMR-
WB) (AMR-WB) speech codec; General
description.", 2015-12, 3GPP TS 26.171 v13.0.0, December 2015,
<http://www.3gpp.org/DynaReport/26171.htm>.

[TS26.173]
3GPP, "3GPP TS 26.173 v13.1.0: ANSI-C "ANSI-C code for the Adaptive Multi-Rate - Wideband
(AMR-WB) speech codec.",
2016-03, codec", 3GPP TS 26.173 v13.1.0, March
2016, <http://www.3gpp.org/DynaReport/26173.htm>.

[TS26.190]
3GPP, "3GPP TS 26.190 v13.0.0: Speech "Speech codec speech processing functions; Adaptive
Multi-Rate - Wideband (AMR-
WB) (AMR-WB) speech codec; Transcoding functions.", 2015-12,
functions", 3GPP TS 26.190 v13.0.0, December 2015,
<http://www.3gpp.org/DynaReport/26190.htm>.

[TS26.204]
3GPP, "3GPP TS 26.204 v13.1.0: Speech "Speech codec speech processing functions; Adaptive
Multi-Rate - Wideband (AMR-
WB) (AMR-WB) speech codec; ANSI-C
code.", 2016-03, 3GPP TS 26.204 v13.1.0, March 2016,
<http://www.3gpp.org/DynaReport/26204.htm>.

8.2.

6.2. Informative references References

[EN300175-1]
ETSI, "ETSI EN 300 175-1, v2.6.1: Digital "Digital Enhanced Cordless Telecommunications
(DECT); Common Interface (CI); Part 1: Overview", ETSI
EN 300 175-1, v2.6.1, 2015, <http://www.etsi.org/deliver/
etsi_en/300100_300199/30017501/02.06.01_60/
en_30017501v020601p.pdf>.

[EN300175-8]
ETSI, "ETSI EN 300 175-8, v2.6.1: Digital "Digital Enhanced Cordless Telecommunications
(DECT); Common Interface (CI); Part 8: Speech and audio
coding and transmission.", ETSI EN 300 175-8, v2.6.1,
2015, <http://www.etsi.org/deliver/
etsi_en/300100_300199/30017508/02.06.01_60/
en_30017508v020601p.pdf>.

[G.711] ITU, "Recommendation ITU-T G.711 (2012): Pulse ITU-T, "Pulse code modulation (PCM) of voice frequencies", 1988-11,
ITU-T Recommendation G.711, November 1988,
<http://www.itu.int/rec/T-REC-G.711-198811-I/en>.

[I-D.ietf-rtcweb-overview]

[IR.36] GSMA, "Adaptive Multirate Wide Band", IR.36, Version 3.0,
September 2014, <http://www.gsma.com/newsroom/all-
documents/
official-document-ir-36-adaptive-multirate-wide-band>.

[OVERVIEW]
Alvestrand, H., "Overview: Real Time Protocols for
Browser-based Applications", draft-ietf-rtcweb-overview-15
(work Work in progress), Progress, draft-ietf-
rtcweb-overview-15, January 2016.

[IR.36] GSMA, "Adaptive Multirate Wide Band V3.0", September 2014,
<http://www.gsma.com/newsroom/all-documents/
official-document-ir-36-adaptive-multirate-wide-band>.

[P.800] ITU, "ITU-T P.800: Methods ITU-T, "Methods for objective and subjective
assessment determination of
transmission quality", 1996-08, <https://www.itu.int/rec/
T-REC-P.800-199608-I/en>. ITU-T Recommendation P.800, August
1996, <https://www.itu.int/rec/T-REC-P.800-199608-I/en>.

[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>.

[TS181005]
ETSI, "Telecommunications and Internet converged Services
and Protocols for Advanced Networking (TISPAN); Service
and Capability Requirements V3.3.1 (2009-12)", ETSI
TS 181005, 2009, <http://www.etsi.org/deliver/
etsi_ts/181000_181099/181005/03.03.01_60/
ts_181005v030301p.pdf>.

[TS23.002]
3GPP, "3GPP TS 23.002 v13.5.0: Network "Network architecture",
2016-03, 3GPP TS23.002 v13.5.0, March
2016, <http://www.3gpp.org/dynareport/23002.htm>.

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