draft-ietf-rtcweb-audio-codecs-for-interop-04.txt   draft-ietf-rtcweb-audio-codecs-for-interop-05.txt 
Network Working Group S. Proust Network Working Group S. Proust, Ed.
Internet-Draft Orange Internet-Draft Orange
Intended status: Informational December 11, 2015 Intended status: Informational February 10, 2016
Expires: June 13, 2016 Expires: August 13, 2016
Additional WebRTC audio codecs for interoperability. Additional WebRTC audio codecs for interoperability.
draft-ietf-rtcweb-audio-codecs-for-interop-04 draft-ietf-rtcweb-audio-codecs-for-interop-05
Abstract Abstract
To ensure a baseline level of interoperability between WebRTC To ensure a baseline level of interoperability between WebRTC
clients, a minimum set of required codecs is specified. However, to endpoints, a minimum set of required codecs is specified. However,
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
considered for WebRTC clients to address the most relevant considered for WebRTC endpoints to address the most relevant
interoperability use cases. interoperability use cases.
Status of This Memo Status of This Memo
This Internet-Draft is submitted in full conformance with the This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/. Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
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 June 13, 2016. This Internet-Draft will expire on August 13, 2016.
Copyright Notice Copyright Notice
Copyright (c) 2015 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
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the Trust Legal Provisions and are provided without warranty as the Trust Legal Provisions and are provided without warranty as
skipping to change at page 2, line 24 skipping to change at page 2, line 24
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 . . . . . . . . . . . . . . . . . . . . . . . 6 WebRTC . . . . . . . . . . . . . . . . . . . . . . . 7
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
4.4. Other codecs . . . . . . . . . . . . . . . . . . . . . . 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
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 11 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 11
1. Introduction 1. Introduction
As indicated in [I-D.ietf-rtcweb-overview], it has been anticipated As indicated in [I-D.ietf-rtcweb-overview], it has been anticipated
that WebRTC will not remain an isolated island and that some WebRTC that WebRTC will not remain an isolated island and that some WebRTC
endpoints will need to communicate with devices used in other endpoints will need to communicate with devices used in other
existing networks with the help of a gateway. Therefore, in order to existing networks with the help of a gateway. Therefore, in order to
maximize the possibility to establish the session without the need maximize the possibility to establish the session without the need
for audio transcoding, it is recommended in [I-D.ietf-rtcweb-audio] for audio transcoding, it is recommended in [I-D.ietf-rtcweb-audio]
to include in the offer other suitable audio codecs that are to include in the offer other suitable audio codecs beyond those that
available to the browser. This document provides some guidelines on are mandatory to implement. This document provides some guidelines
the suitable codecs to be considered for WebRTC clients to address on the suitable codecs to be considered for WebRTC endpoints to
the most relevant interoperability use cases. address the most relevant interoperability use cases.
The codecs considered in this document are recommended to be The codecs considered in this document are recommended to be
supported and included in the Offer only for WebRTC clients for which supported and included in the Offer only for WebRTC endpoints for
interoperability with other non-WebRTC endpoints and non-WebRTC based which interoperability with other non-WebRTC endpoints and non-WebRTC
services is relevant as described in Section 4.1.2, Section 4.2.2, based services is relevant as described in Section 4.1.2,
Section 4.3.2. Other use cases may justify offering other additional Section 4.2.2, Section 4.3.2. Other use cases may justify offering
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
WebRTC non browser (also called "WebRTC device" or "WebRTC native
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
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o MOS: Mean Opinion Score o MOS: Mean Opinion Score
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 clients can The mandatory implementation of OPUS [RFC6716] in WebRTC endpoints
guarantee codec interoperability (without transcoding) at state of can guarantee codec interoperability (without transcoding) at state
the art voice quality (better than narrow band "PSTN" quality) of the art voice quality (better than narrow band "PSTN" quality)
between WebRTC clients. The WebRTC technology is also expected to be between WebRTC endpoints. The WebRTC technology is also expected to
used to communicate with other types of clients 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 clients 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, DECT terminals that do not support WebRTC
clients nor implement OPUS. As a consequence, these calls are likely endpoints nor implement OPUS. As a consequence, these calls are
to be either of low narrow band PSTN quality using G.711 [G.711] at likely to be either of low narrow band PSTN quality using G.711
both ends or affected by transcoding operations. The drawback of [G.711] at both ends or affected by transcoding operations. The
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
stressed that if G.711 is used as a fall back codec for stressed that if G.711 is used as a fall back codec for
interoperation, wideband voice quality will be lost. Such interoperation, wideband voice quality will be lost. Such
bandwidth reduction effect down to narrow band clearly degrades bandwidth reduction effect down to narrow band clearly degrades
the user perceived quality of service leading to shorter and less the user perceived quality of service leading to shorter and less
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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 especially 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 implemented in several hundreds of device terminals. AMR-WB is now very widely implemented in devices and
models and 145 HD mobile networks in 85 countries with a customer networks offering "HD Voice" A high number of calls are consequently
base of more than 450 million. A high number of calls are likely to occur between WebRTC endpoints and mobile 3GPP terminals
consequently likely to occur between WebRTC clients and mobile 3GPP offering AMR-WB. The use of AMR-WB by WebRTC endpoints would
terminals. The use of AMR-WB by WebRTC clients would consequently consequently allow transcoding free interoperation with all mobile
allow transcoding free interoperation with all mobile 3GPP wideband 3GPP wideband terminals. Besides, WebRTC endpoints running on mobile
terminals. Besides, WebRTC clients running on mobile terminals terminals (smartphones) may reuse the AMR-WB codec already
(smartphones) may reuse the AMR-WB codec already implemented on these implemented on these devices.
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 packets. 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
client offering AMR-WB it is important that the WebRTC client: 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
Efficient payload format preferred. Efficient payload format preferred.
o Be capable of operating AMR-WB with any subset of the nine codec o Be capable of operating AMR-WB with any subset of the nine codec
modes and source controlled rate operation. Offer at least one modes and source controlled rate operation. Offer at least one
AMR-WB configuration with parameter settings as defined in AMR-WB configuration with parameter settings as defined in
Table 6.1 of [TS26.114]. In order to maximize the Table 6.1 of [TS26.114]. In order to maximize the
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, i.e., several hundred millions of terminals. This communication. This include both mobile phone calls using GSM and 3G
include both mobile phone calls using GSM and 3G cellular systems as cellular systems as well as multimedia telephony services over IP/IMS
well as multimedia telephony services over IP/IMS and 4G/VoLTE, such and 4G/VoLTE, such as, GSMA voice IMS profile for VoLTE in [IR.92].
as, GSMA voice IMS profile for VoLTE in [IR.92]. In addition to In addition to impacts listed above, support of AMR can avoid
impacts listed above, support of AMR can avoid degrading the high degrading the high efficiency over mobile radio access.References for
efficiency over mobile radio access.References for AMR related AMR related specifications including detailed codec description and
specifications including detailed codec description and source code source code are in [TS26.071], [TS26.073], [TS26.090], [TS26.104].
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 client 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 client running on mobile terminals (smartphones) may Besides, WebRTC endpoints running on mobile terminals (smartphones)
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 packets. 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
client offering AMR, it is important that the WebRTC client: 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.
o Offer at least one configuration with parameter settings as 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 defined in Table 6.1 and Table 6.2 of [TS26.114]. In order to
maximize the interoperability and quality this offer shall not maximize the interoperability and quality this offer shall not
restrict AMR codec modes offered. Restrictions in the use of restrict AMR codec modes offered. Restrictions in the use of
codec modes may be included in the answer. codec modes may be included in the answer.
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wideband codec for New Generation DECT with purpose to greatly wideband codec for New Generation DECT with purpose to greatly
increase the voice quality by extending the bandwidth from narrow increase the voice quality by extending the bandwidth from narrow
band to wideband. G.722 is the wideband codec required for CAT-iq band to wideband. G.722 is the wideband codec required for CAT-iq
DECT certified terminals and the V2.0 of CAT-iq specifications have DECT certified terminals and the V2.0 of CAT-iq specifications have
been approved by GSMA as minimum requirements for HD voice logo usage been approved by GSMA as minimum requirements for HD voice logo usage
on "fixed" devices; i.e., broadband connections using the G.722 on "fixed" devices; i.e., broadband connections using the G.722
codec. codec.
4.3.2. WebRTC relevant use case for G.722 4.3.2. WebRTC relevant use case for G.722
G.722 is the wideband codec required for DECT CAT-iq terminals. The G.722 is the wideband codec required for DECT CAT-iq terminals. DECT
market for DECT cordless phones including DECT chipset is more than cordeless phones are still widely used to offer short range wireless
150 million per year and CAT-IQ is a registered trade make in 47 connection to PSTN or VoIP services. G.722 has also been specified
countries worldwide. G.722 has also been specified by ETSI in by ETSI in [TS181005] as mandatory wideband codec for IMS multimedia
[TS181005] as mandatory wideband codec for IMS multimedia telephony telephony communication service and supplementary services using
communication service and supplementary services using fixed fixed broadband access. The support of G.722 would consequently
broadband access. The support of G.722 would consequently allow allow transcoding free IP interoperation between WebRTC endpoints and
transcoding free IP interoperation between WebRTC client and fixed fixed VoIP terminals including DECT / CAT-IQ terminals supporting
VoIP terminals including DECT / CAT-IQ terminals supporting G.722. G.722. Besides, WebRTC endpoints running on fixed terminals
Besides, WebRTC client running on fixed terminals implementing G.722 implementing G.722 may reuse the G.722 codec already implemented on
may reuse the G.722 codec already implemented on these devices. these devices.
4.3.3. Guidelines for G.722 usage and implementation 4.3.3. Guidelines for G.722 usage and implementation
The payload format to be used for G.722 is defined in [RFC3551] with 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- each octet of the stream of octets produced by the codec to be octet-
aligned in an RTP packet. The sampling frequency for G.722 is 16 kHz aligned in an RTP packet. The sampling frequency for G.722 is 16 kHz
but the rtp clock rate is set to 8000Hz in SDP to stay backward but the rtp clock rate is set to 8000Hz in SDP to stay backward
compatible with an erroneous definition in the original version of compatible with an erroneous definition in the original version of
the RTP A/V profile. Further guidelines for implementing and using the RTP A/V profile. Further guidelines for implementing and using
G.722 with purpose to ensure interoperability with multimedia G.722 with purpose to ensure interoperability with multimedia
telephony services over IMS can be found in section 7 of [TS26.114]. telephony services over IMS can be found in section 7 of [TS26.114].
Additional information of G.722 implementation in DECT can be found Additional information of G.722 implementation in DECT can be found
in [EN300175-8] and full codec description and C source code in in [EN300175-8] and full codec description and C source code in
[G.722]. [G.722].
4.4. Other codecs
Other interoperability use cases may justify the use of other codecs.
5. Security Considerations 5. Security Considerations
Security considerations for WebRTC Audio Codec and Processing Security considerations for WebRTC Audio Codec and Processing
Requirements can be found in [I-D.ietf-rtcweb-audio]. Implementors Requirements can be found in [I-D.ietf-rtcweb-audio]. Implementors
making use of the additional codecs considered in this document are making use of the additional codecs considered in this document are
advised to also report more specifically to the "Security advised to also refer more specifically to the "Security
Considerations" sections of [RFC4867] (for AMR and AMR-WB) and Considerations" sections of [RFC4867] (for AMR and AMR-WB) and
[RFC3551]. [RFC3551].
6. IANA Considerations 6. IANA Considerations
None. None.
7. Acknowledgements 7. Acknowledgements
The authors of this document are The authors of this document are
skipping to change at page 11, line 7 skipping to change at page 11, line 7
[EN300175-8] [EN300175-8]
ETSI, "ETSI EN 300 175-8, v2.5.1: Digital Enhanced ETSI, "ETSI EN 300 175-8, v2.5.1: 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.", 2009.
[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.
[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-14 Browser-based Applications", draft-ietf-rtcweb-overview-15
(work in progress), June 2015. (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.
[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.
[TS23.002] [TS23.002]
3GPP, "3GPP TS 23.002 v13.3.0: Network architecture", 3GPP, "3GPP TS 23.002 v13.3.0: Network architecture",
2015-09. 2015-09.
Author's Address Author's Address
Stephane Proust 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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