draft-ietf-rtcweb-audio-codecs-for-interop-05.txt   draft-ietf-rtcweb-audio-codecs-for-interop-06.txt 
Network Working Group S. Proust, Ed. Network Working Group S. Proust, Ed.
Internet-Draft Orange Internet-Draft Orange
Intended status: Informational February 10, 2016 Intended status: Informational April 22, 2016
Expires: August 13, 2016 Expires: October 24, 2016
Additional WebRTC audio codecs for interoperability. Additional WebRTC audio codecs for interoperability.
draft-ietf-rtcweb-audio-codecs-for-interop-05 draft-ietf-rtcweb-audio-codecs-for-interop-06
Abstract Abstract
To ensure a baseline level of interoperability between WebRTC To ensure a baseline level of interoperability between WebRTC
endpoints, a minimum set of required codecs is specified. However, endpoints, a minimum set of required codecs is specified. However,
to maximize the possibility to establish the session without the need to maximize the possibility to establish the session without the need
for audio transcoding, it is also recommended to include in the offer for audio transcoding, it is also recommended to include in the offer
other suitable audio codecs that are available to the browser. other suitable audio codecs that are available to the browser.
This document provides some guidelines on the suitable codecs to be This document provides some guidelines on the suitable codecs to be
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Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
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time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on August 13, 2016. This Internet-Draft will expire on October 24, 2016.
Copyright Notice Copyright Notice
Copyright (c) 2016 IETF Trust and the persons identified as the Copyright (c) 2016 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of (http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents publication of this document. Please review these documents
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4. Additional suitable codecs for WebRTC . . . . . . . . . . . . 5 4. Additional suitable codecs for WebRTC . . . . . . . . . . . . 5
4.1. AMR-WB . . . . . . . . . . . . . . . . . . . . . . . . . 5 4.1. AMR-WB . . . . . . . . . . . . . . . . . . . . . . . . . 5
4.1.1. AMR-WB General description . . . . . . . . . . . . . 5 4.1.1. AMR-WB General description . . . . . . . . . . . . . 5
4.1.2. WebRTC relevant use case for AMR-WB . . . . . . . . . 5 4.1.2. WebRTC relevant use case for AMR-WB . . . . . . . . . 5
4.1.3. Guidelines for AMR-WB usage and implementation with 4.1.3. Guidelines for AMR-WB usage and implementation with
WebRTC . . . . . . . . . . . . . . . . . . . . . . . 5 WebRTC . . . . . . . . . . . . . . . . . . . . . . . 5
4.2. AMR . . . . . . . . . . . . . . . . . . . . . . . . . . . 6 4.2. AMR . . . . . . . . . . . . . . . . . . . . . . . . . . . 6
4.2.1. AMR General description . . . . . . . . . . . . . . . 6 4.2.1. AMR General description . . . . . . . . . . . . . . . 6
4.2.2. WebRTC relevant use case for AMR . . . . . . . . . . 6 4.2.2. WebRTC relevant use case for AMR . . . . . . . . . . 6
4.2.3. Guidelines for AMR usage and implementation with 4.2.3. Guidelines for AMR usage and implementation with
WebRTC . . . . . . . . . . . . . . . . . . . . . . . 7 WebRTC . . . . . . . . . . . . . . . . . . . . . . . 6
4.3. G.722 . . . . . . . . . . . . . . . . . . . . . . . . . . 7 4.3. G.722 . . . . . . . . . . . . . . . . . . . . . . . . . . 7
4.3.1. G.722 General description . . . . . . . . . . . . . . 7 4.3.1. G.722 General description . . . . . . . . . . . . . . 7
4.3.2. WebRTC relevant use case for G.722 . . . . . . . . . 7 4.3.2. WebRTC relevant use case for G.722 . . . . . . . . . 7
4.3.3. Guidelines for G.722 usage and implementation . . . . 8 4.3.3. Guidelines for G.722 usage and implementation . . . . 8
5. Security Considerations . . . . . . . . . . . . . . . . . . . 8 5. Security Considerations . . . . . . . . . . . . . . . . . . . 8
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8 6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 8 7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 8
8. References . . . . . . . . . . . . . . . . . . . . . . . . . 9 8. References . . . . . . . . . . . . . . . . . . . . . . . . . 9
8.1. Normative references . . . . . . . . . . . . . . . . . . 9 8.1. Normative references . . . . . . . . . . . . . . . . . . 9
8.2. Informative references . . . . . . . . . . . . . . . . . 10 8.2. Informative references . . . . . . . . . . . . . . . . . 10
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which interoperability with other non-WebRTC endpoints and non-WebRTC which interoperability with other non-WebRTC endpoints and non-WebRTC
based services is relevant as described in Section 4.1.2, 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 Section 4.2.2, Section 4.3.2. Other use cases may justify offering
other additional codecs to avoid transcoding. other additional codecs to avoid transcoding.
2. Definition and abbreviations 2. Definition and abbreviations
o Legacy networks: In this document, legacy networks encompass the o Legacy networks: In this document, legacy networks encompass the
conversational networks that are already deployed like the PSTN, conversational networks that are already deployed like the PSTN,
the PLMN, the IP/IMS networks offering VoIP services, including the PLMN, the IP/IMS networks offering VoIP services, including
3GPP "4G" Evolved Packet System[TS23.002] supporting voice over 3GPP "4G" Evolved Packet System [TS23.002] supporting voice over
LTE radio access (VoLTE) [IR.92]. LTE radio access (VoLTE) [IR.92].
o WebRTC endpoint: a WebRTC endpoint can be a WebRTC browser or a o WebRTC endpoint: a WebRTC endpoint can be a WebRTC browser or a
WebRTC non browser (also called "WebRTC device" or "WebRTC native WebRTC non browser (also called "WebRTC device" or "WebRTC native
application") as defined in [I-D.ietf-rtcweb-overview] application") as defined in [I-D.ietf-rtcweb-overview]
o AMR: Adaptive Multi-Rate. o AMR: Adaptive Multi-Rate.
o AMR-WB: Adaptive Multi-Rate WideBand. o AMR-WB: Adaptive Multi-Rate WideBand.
o CAT-iq: Cordless Advanced Technology-internet and quality. o CAT-iq: Cordless Advanced Technology-internet and quality.
o DECT: Digital Enhanced Cordless Telecommunications o DECT: Digital Enhanced Cordless Telecommunications
o IMS: IP Multimedia Subsystem o IMS: IP Multimedia Subsystem
o LTE: Long Term Evolution (3GPP "4G" wireless data transmission o LTE: Long Term Evolution (3GPP "4G" wireless data transmission
standard) standard)
o MOS: Mean Opinion Score o MOS: Mean Opinion Score, defined in ITU-T P.800 specification
[P.800]
o PSTN:Public Switched Telephone Network o PSTN: Public Switched Telephone Network
o PLMN: Public Land Mobile Network o PLMN: Public Land Mobile Network
o VoLTE: Voice Over LTE o VoLTE: Voice Over LTE
3. Rationale for additional WebRTC codecs 3. Rationale for additional WebRTC codecs
The mandatory implementation of OPUS [RFC6716] in WebRTC endpoints The mandatory implementation of OPUS [RFC6716] in WebRTC endpoints
can guarantee codec interoperability (without transcoding) at state can guarantee codec interoperability (without transcoding) at state
of the art voice quality (better than narrow band "PSTN" quality) of the art voice quality (better than narrow band "PSTN" quality)
between WebRTC endpoints. The WebRTC technology is also expected to between WebRTC endpoints. The WebRTC technology is also expected to
be used to communicate with other types of endpoints using other be used to communicate with other types of endpoints using other
technologies. It can be used for instance as an access technology to technologies. It can be used for instance as an access technology to
VoLTE services (Voice over LTE as specified in [IR.92]) or to VoLTE services (Voice over LTE as specified in [IR.92]) or to
interoperate with fixed or mobile Circuit Switched or VoIP services interoperate with fixed or mobile Circuit Switched or VoIP services
like mobile Circuit Switched voice over 3GPP 2G/3G mobile networks like mobile Circuit Switched voice over 3GPP 2G/3G mobile networks
[TS23.002] or DECT based VoIP telephony [EN300175-1]. Consequently, [TS23.002] or DECT based VoIP telephony [EN300175-1]. Consequently,
a significant number of calls are likely to occur between terminals a significant number of calls are likely to occur between terminals
supporting WebRTC endpoints and other terminals like mobile handsets, supporting WebRTC endpoints and other terminals like mobile handsets,
fixed VoIP terminals, DECT terminals that do not support WebRTC fixed VoIP terminals and DECT terminals that do not support WebRTC
endpoints nor implement OPUS. As a consequence, these calls are endpoints nor implement OPUS. As a consequence, these calls are
likely to be either of low narrow band PSTN quality using G.711 likely to be either of low narrow band PSTN quality using G.711
[G.711] at both ends or affected by transcoding operations. The [G.711] at both ends or affected by transcoding operations. The
drawback of such transcoding operations are listed below: drawback of such transcoding operations are listed below:
o Degraded user experience with respect to voice quality: voice o Degraded user experience with respect to voice quality: voice
quality is significantly degraded by transcoding. For instance, quality is significantly degraded by transcoding. For instance,
the degradation is around 0.2 to 0.3 MOS for most of transcoding 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 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 the same range for other wideband transcoding cases. It should be
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according to the foreseen interoperability cases to be addressed. according to the foreseen interoperability cases to be addressed.
4.1. AMR-WB 4.1. AMR-WB
4.1.1. AMR-WB General description 4.1.1. AMR-WB General description
The Adaptive Multi-Rate WideBand (AMR-WB) is a 3GPP defined speech The Adaptive Multi-Rate WideBand (AMR-WB) is a 3GPP defined speech
codec that is mandatory to implement in any 3GPP terminal that codec that is mandatory to implement in any 3GPP terminal that
supports wideband speech communication. It is being used in circuit supports wideband speech communication. It is being used in circuit
switched mobile telephony services and new multimedia telephony switched mobile telephony services and new multimedia telephony
services over IP/IMS. It is especially used for voice over LTE as 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 specified by GSMA in [IR.92]. More detailed information on AMR-WB
can be found in [IR.36]. References for AMR-WB related can be found in [IR.36]. References for AMR-WB related
specifications including detailed codec description and source code specifications including detailed codec description and source code
are in [TS26.171], [TS26.173], [TS26.190], [TS26.204]. are in [TS26.171], [TS26.173], [TS26.190], [TS26.204].
4.1.2. WebRTC relevant use case for AMR-WB 4.1.2. WebRTC relevant use case for AMR-WB
The market of personal voice communication is driven by mobile The market of personal voice communication is driven by mobile
terminals. AMR-WB is now very widely implemented in devices and terminals. AMR-WB is now very widely implemented in devices and
networks offering "HD Voice" A high number of calls are consequently networks offering "HD Voice" A high number of calls are consequently
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offering AMR-WB. The use of AMR-WB by WebRTC endpoints would offering AMR-WB. The use of AMR-WB by WebRTC endpoints would
consequently allow transcoding free interoperation with all mobile consequently allow transcoding free interoperation with all mobile
3GPP wideband terminals. Besides, WebRTC endpoints running on mobile 3GPP wideband terminals. Besides, WebRTC endpoints running on mobile
terminals (smartphones) may reuse the AMR-WB codec already terminals (smartphones) may reuse the AMR-WB codec already
implemented on these devices. implemented on these devices.
4.1.3. Guidelines for AMR-WB usage and implementation with WebRTC 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] The payload format to be used for AMR-WB is described in [RFC4867]
with bandwidth efficient format and one speech frame encapsulated in with bandwidth efficient format and one speech frame encapsulated in
each RTP packets. Further guidelines for implementing and using AMR- each RTP packet. Further guidelines for implementing and using AMR-
WB and ensuring interoperability with 3GPP mobile services can be WB and ensuring interoperability with 3GPP mobile services can be
found in [TS26.114]. In order to ensure interoperability with 4G/ found in [TS26.114]. In order to ensure interoperability with 4G/
VoLTE as specified by GSMA, the more specific IMS profile for voice 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 derived from [TS26.114] should be considered in [IR.92]. In order to
maximize the possibility of successful call establishment for WebRTC maximize the possibility of successful call establishment for WebRTC
endpoints offering AMR-WB it is important that the WebRTC endpoints: 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 o Offer AMR in addition to AMR-WB with AMR-WB listed first (AMR-WB
being a wideband codec) as preferred payload type with respect to being a wideband codec) as preferred payload type with respect to
other narrow band codecs (AMR, G.711...) and with Bandwidth other narrow band codecs (AMR, G.711...) and with Bandwidth
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interoperability and quality this offer does not restrict the interoperability and quality this offer does not restrict the
codec modes offered. Restrictions in the use of codec modes may codec modes offered. Restrictions in the use of codec modes may
be included in the answer. be included in the answer.
4.2. AMR 4.2. AMR
4.2.1. AMR General description 4.2.1. AMR General description
Adaptive Multi-Rate (AMR) is a 3GPP defined speech codec that is Adaptive Multi-Rate (AMR) is a 3GPP defined speech codec that is
mandatory to implement in any 3GPP terminal that supports voice mandatory to implement in any 3GPP terminal that supports voice
communication. This include both mobile phone calls using GSM and 3G communication. This includes both mobile phone calls using GSM and
cellular systems as well as multimedia telephony services over IP/IMS 3G cellular systems as well as multimedia telephony services over IP/
and 4G/VoLTE, such as, GSMA voice IMS profile for VoLTE in [IR.92]. IMS and 4G/VoLTE, such as GSMA voice IMS profile for VoLTE in
In addition to impacts listed above, support of AMR can avoid [IR.92]. In addition to impacts listed above, support of AMR can
degrading the high efficiency over mobile radio access.References for avoid degrading the high efficiency over mobile radio access.
AMR related specifications including detailed codec description and References for AMR related specifications including detailed codec
source code are in [TS26.071], [TS26.073], [TS26.090], [TS26.104]. description and source code are in [TS26.071], [TS26.073],
[TS26.090], [TS26.104].
4.2.2. WebRTC relevant use case for AMR 4.2.2. WebRTC relevant use case for AMR
A user of a WebRTC endpoint on a device integrating an AMR module 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 wants to communicate with another user that can only be reached on a
mobile device that only supports AMR. Although more and more mobile device that only supports AMR. Although more and more
terminal devices are now "HD voice" and support AMR-WB; there are terminal devices are now "HD voice" and support AMR-WB; there are
still a high number of legacy terminals supporting only AMR still a high number of legacy terminals supporting only AMR
(terminals with no wideband / HD Voice capabilities) that are still (terminals with no wideband / HD Voice capabilities) that are still
in use. The use of AMR by WebRTC endpoints would consequently allow in use. The use of AMR by WebRTC endpoints would consequently allow
transcoding free interoperation with all mobile 3GPP terminals. transcoding free interoperation with all mobile 3GPP terminals.
Besides, WebRTC endpoints running on mobile terminals (smartphones) Besides, WebRTC endpoints running on mobile terminals (smartphones)
may reuse the AMR codec already implemented on these devices. may reuse the AMR codec already implemented on these devices.
4.2.3. Guidelines for AMR usage and implementation with WebRTC 4.2.3. Guidelines for AMR usage and implementation with WebRTC
The payload format to be used for AMR is described in [RFC4867] with The payload format to be used for AMR is described in [RFC4867] with
bandwidth efficient format and one speech frame encapsulated in each bandwidth efficient format and one speech frame encapsulated in each
RTP packets. Further guidelines for implementing and using AMR with RTP packet. Further guidelines for implementing and using AMR with
purpose to ensure interoperability with 3GPP mobile services can be purpose to ensure interoperability with 3GPP mobile services can be
found in [TS26.114]. In order to ensure interoperability with 4G/ found in [TS26.114]. In order to ensure interoperability with 4G/
VoLTE as specified by GSMA, the more specific IMS profile for voice 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 derived from [TS26.114] should be considered in [IR.92]. In order to
maximize the possibility of successful call establishment for WebRTC maximize the possibility of successful call establishment for WebRTC
endpoints offering AMR, it is important that the WebRTC endpoints: endpoints offering AMR, it is important that the WebRTC endpoints:
o Be capable of operating AMR with any subset of the eight codec o Be capable of operating AMR with any subset of the eight codec
modes and source controlled rate operation. modes and source controlled rate operation.
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The authors would like to thank Magnus Westerlund, Barry Dingle and The authors would like to thank Magnus Westerlund, Barry Dingle and
Sanjay Mishra who carefully reviewed the document and helped to Sanjay Mishra who carefully reviewed the document and helped to
improve it. improve it.
8. References 8. References
8.1. Normative references 8.1. Normative references
[G.722] ITU, "Recommendation ITU-T G.722 (2012): 7 kHz audio- [G.722] ITU, "Recommendation ITU-T G.722 (2012): 7 kHz audio-
coding within 64 kbit/s", 2012-09. coding within 64 kbit/s", 2012-09,
<http://www.itu.int/rec/T-REC-G.722-201209-I/en>.
[I-D.ietf-rtcweb-audio] [I-D.ietf-rtcweb-audio]
Valin, J. and C. Bran, "WebRTC Audio Codec and Processing Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
Requirements", draft-ietf-rtcweb-audio-09 (work in Requirements", draft-ietf-rtcweb-audio-10 (work in
progress), November 2015. progress), February 2016.
[IR.92] GSMA, "IMS Profile for Voice and SMS V9.0", April 2015. [IR.92] GSMA, "IMS Profile for Voice and SMS V9.0", 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 [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551, Video Conferences with Minimal Control", STD 65, RFC 3551,
DOI 10.17487/RFC3551, July 2003, DOI 10.17487/RFC3551, July 2003,
<http://www.rfc-editor.org/info/rfc3551>. <http://www.rfc-editor.org/info/rfc3551>.
[RFC4867] Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q. Xie, [RFC4867] Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q. Xie,
"RTP Payload Format and File Storage Format for the "RTP Payload Format and File Storage Format for the
Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband
(AMR-WB) Audio Codecs", RFC 4867, DOI 10.17487/RFC4867, (AMR-WB) Audio Codecs", RFC 4867, DOI 10.17487/RFC4867,
April 2007, <http://www.rfc-editor.org/info/rfc4867>. April 2007, <http://www.rfc-editor.org/info/rfc4867>.
[TS26.071] [TS26.071]
3GPP, "3GPP TS 26.071 v12.0.0: Recommendation ITU-T G.722 3GPP, "3GPP TS 26.071 v13.0.0: Recommendation ITU-T G.722
(2012): "Mandatory Speech Codec speech processing (2012): "Mandatory Speech Codec speech processing
functions; AMR Speech CODEC; General description".", functions; AMR Speech CODEC; General description".",
2014-09. 2015-12, <http://www.3gpp.org/DynaReport/26071.htm>.
[TS26.073] [TS26.073]
3GPP, "3GPP TS 26.073 v12.0.0: ANSI C code for the 3GPP, "3GPP TS 26.073 v13.0.0: ANSI C code for the
Adaptive Multi Rate (AMR) speech codec", 2014-09. Adaptive Multi Rate (AMR) speech codec", 2015-12,
<http://www.3gpp.org/DynaReport/26073.htm>.
[TS26.090] [TS26.090]
3GPP, "3GPP TS 26.090 v12.0.0: Mandatory Speech Codec 3GPP, "3GPP TS 26.090 v13.0.0: Mandatory Speech Codec
speech processing functions; Adaptive Multi-Rate (AMR) speech processing functions; Adaptive Multi-Rate (AMR)
speech codec; Transcoding functions.", 2014-09. speech codec; Transcoding functions.", 2015-12,
<http://www.3gpp.org/DynaReport/26090.htm>.
[TS26.104] [TS26.104]
3GPP, "3GPP TS 26.104 v12.0.0: ANSI C code for the 3GPP, "3GPP TS 26.104 v13.0.0: ANSI C code for the
floating-point Adaptive Multi Rate (AMR) speech codec.", floating-point Adaptive Multi Rate (AMR) speech codec.",
2014-09. 2015-12, <http://www.3gpp.org/DynaReport/26090.htm>.
[TS26.114] [TS26.114]
3GPP, "IP Multimedia Subsystem (IMS); Multimedia 3GPP, "3GPP TS 26.114 v13.3.0: IP Multimedia Subsystem
telephony; Media handling and interaction V13.0.0", June (IMS); Multimedia telephony; Media handling and
2015. interaction", March 2016,
<http://www.3gpp.org/DynaReport/26114.htm>.
[TS26.171] [TS26.171]
3GPP, "3GPP TS 26.071 v12.0.0: Recommendation ITU-T G.722 3GPP, "3GPP TS 26.171 v13.0.0: Speech codec speech
(2012): "Speech codec speech processing functions; processing functions; Adaptive Multi-Rate - Wideband (AMR-
Adaptive Multi-Rate - Wideband (AMR-WB) speech codec; WB) speech codec; General description.", 2015-12,
General description".", 2014-09. <http://www.3gpp.org/DynaReport/26171.htm>.
[TS26.173] [TS26.173]
3GPP, "3GPP TS 26.073 v12.1.0: ANSI-C code for the 3GPP, "3GPP TS 26.173 v13.1.0: ANSI-C code for the
Adaptive Multi-Rate - Wideband (AMR-WB) speech codec.", Adaptive Multi-Rate - Wideband (AMR-WB) speech codec.",
2015-03. 2016-03, <http://www.3gpp.org/DynaReport/26173.htm>.
[TS26.190] [TS26.190]
3GPP, "3GPP TS 26.090 v12.0.0: Speech codec speech 3GPP, "3GPP TS 26.190 v13.0.0: Speech codec speech
processing functions; Adaptive Multi-Rate - Wideband (AMR- processing functions; Adaptive Multi-Rate - Wideband (AMR-
WB) speech codec; Transcoding functions.", 2014-09. WB) speech codec; Transcoding functions.", 2015-12,
<http://www.3gpp.org/DynaReport/26190.htm>.
[TS26.204] [TS26.204]
3GPP, "3GPP TS 26.104 v12.1.0: Speech codec speech 3GPP, "3GPP TS 26.204 v13.1.0: Speech codec speech
processing functions; Adaptive Multi-Rate - Wideband (AMR- processing functions; Adaptive Multi-Rate - Wideband (AMR-
WB) speech codec; ANSI-C code.", 2015-03. WB) speech codec; ANSI-C code.", 2016-03,
<http://www.3gpp.org/DynaReport/26204.htm>.
8.2. Informative references 8.2. Informative references
[EN300175-1] [EN300175-1]
ETSI, "ETSI EN 300 175-1, Digital Enhanced Cordless ETSI, "ETSI EN 300 175-1, v2.6.1: Digital Enhanced
Telecommunications (DECT); Common Interface (CI); Part 1: Cordless Telecommunications (DECT); Common Interface (CI);
Overview v2.5.1", 2009. Part 1: Overview", 2015, <http://www.etsi.org/deliver/
etsi_en/300100_300199/30017501/02.06.01_60/
en_30017501v020601p.pdf>.
[EN300175-8] [EN300175-8]
ETSI, "ETSI EN 300 175-8, v2.5.1: Digital Enhanced ETSI, "ETSI EN 300 175-8, v2.6.1: Digital Enhanced
Cordless Telecommunications (DECT); Common Interface (CI); Cordless Telecommunications (DECT); Common Interface (CI);
Part 8: Speech and audio coding and transmission.", 2009. Part 8: Speech and audio coding and transmission.", 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 code [G.711] ITU, "Recommendation ITU-T G.711 (2012): Pulse code
modulation (PCM) of voice frequencies", 1988-11. modulation (PCM) of voice frequencies", 1988-11,
<http://www.itu.int/rec/T-REC-G.711-198811-I/en>.
[I-D.ietf-rtcweb-overview] [I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for Alvestrand, H., "Overview: Real Time Protocols for
Browser-based Applications", draft-ietf-rtcweb-overview-15 Browser-based Applications", draft-ietf-rtcweb-overview-15
(work in progress), January 2016. (work in progress), January 2016.
[IR.36] GSMA, "Adaptive Multirate Wide Band V3.0", September 2014. [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 for objective and subjective
assessment of quality", 1996-08, <https://www.itu.int/rec/
T-REC-P.800-199608-I/en>.
[RFC6716] Valin, JM., Vos, K., and T. Terriberry, "Definition of the [RFC6716] Valin, JM., Vos, K., and T. Terriberry, "Definition of the
Opus Audio Codec", RFC 6716, DOI 10.17487/RFC6716, Opus Audio Codec", RFC 6716, DOI 10.17487/RFC6716,
September 2012, <http://www.rfc-editor.org/info/rfc6716>. September 2012, <http://www.rfc-editor.org/info/rfc6716>.
[TS181005] [TS181005]
ETSI, "Telecommunications and Internet converged Services ETSI, "Telecommunications and Internet converged Services
and Protocols for Advanced Networking (TISPAN); Service and Protocols for Advanced Networking (TISPAN); Service
and Capability Requirements V3.3.1 (2009-12)", 2009. and Capability Requirements V3.3.1 (2009-12)", 2009,
<http://www.etsi.org/deliver/
etsi_ts/181000_181099/181005/03.03.01_60/
ts_181005v030301p.pdf>.
[TS23.002] [TS23.002]
3GPP, "3GPP TS 23.002 v13.3.0: Network architecture", 3GPP, "3GPP TS 23.002 v13.5.0: Network architecture",
2015-09. 2016-03, <http://www.3gpp.org/dynareport/23002.htm>.
Author's Address Author's Address
Stephane Proust (editor) Stephane Proust (editor)
Orange Orange
2, avenue Pierre Marzin 2, avenue Pierre Marzin
Lannion 22307 Lannion 22307
France France
Email: stephane.proust@orange.com Email: stephane.proust@orange.com
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