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Versions: (draft-marjou-rtcweb-audio-codecs-for-interop) 00 01 02 03 04 05 06 RFC 7875

Network Working Group                                          S. Proust
Internet-Draft                                                    Orange
Intended status: Informational                                 E. Berger
Expires: July 20, 2015                                             Cisco
                                                               B. Feiten
                                                        Deutsche Telekom
                                                               B. Burman
                                                                Ericsson
                                                             K. Bogineni
                                                        Verizon Wireless
                                                                  M. Lei
                                                                  Huawei
                                                              E. Marocco
                                                          Telecom Italia
                                                        January 16, 2015


          Additional WebRTC audio codecs for interoperability.
             draft-ietf-rtcweb-audio-codecs-for-interop-01

Abstract

   To ensure a baseline level of interoperability between WebRTC
   clients, [I-D.ietf-rtcweb-audio] requires a minimum set of codecs.
   However, to maximize the possibility to establish the session without
   the need for audio transcoding, it is also recommended to include in
   the offer other suitable audio codecs that are available to the
   browser.

   This document provides some guidelines on the suitable codecs to be
   considered for WebRTC clients to address the most relevant
   interoperability use cases.

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."




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   This Internet-Draft will expire on July 20, 2015.

Copyright Notice

   Copyright (c) 2015 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
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   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Definitions . . . . . . . . . . . . . . . . . . . . . . . . .   3
   3.  Rationale for additional WebRTC codecs  . . . . . . . . . . .   3
   4.  Additional suitable codecs for WebRTC . . . . . . . . . . . .   4
     4.1.  AMR-WB  . . . . . . . . . . . . . . . . . . . . . . . . .   5
       4.1.1.  AMR-WB General description  . . . . . . . . . . . . .   5
       4.1.2.  WebRTC relevant use case for AMR-WB . . . . . . . . .   5
       4.1.3.  Guidelines for AMR-WB usage and implementation with
               WebRTC  . . . . . . . . . . . . . . . . . . . . . . .   5
     4.2.  AMR . . . . . . . . . . . . . . . . . . . . . . . . . . .   6
       4.2.1.  AMR General description . . . . . . . . . . . . . . .   6
       4.2.2.  WebRTC relevant use case for AMR  . . . . . . . . . .   6
       4.2.3.  Guidelines for AMR usage and implementation with
               WebRTC  . . . . . . . . . . . . . . . . . . . . . . .   6
     4.3.  G.722 . . . . . . . . . . . . . . . . . . . . . . . . . .   7
       4.3.1.  G.722 General description . . . . . . . . . . . . . .   7
       4.3.2.  WebRTC relevant use case for G.722  . . . . . . . . .   7
       4.3.3.  Guidelines for G.722 usage and implementation . . . .   7
     4.4.  Other codecs  . . . . . . . . . . . . . . . . . . . . . .   7
   5.  Security Considerations . . . . . . . . . . . . . . . . . . .   8
   6.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   8
   7.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .   8
   8.  References  . . . . . . . . . . . . . . . . . . . . . . . . .   8
     8.1.  Normative references  . . . . . . . . . . . . . . . . . .   8
     8.2.  Informative references  . . . . . . . . . . . . . . . . .   8
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .   9






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1.  Introduction

   As indicated in [I-D.ietf-rtcweb-overview], it has been anticipated
   that WebRTC will not remain an isolated island and that some WebRTC
   endpoints will need to communicate with devices used in other
   existing networks with the help of a gateway.  Therefore, in order to
   maximize the possibility to establish the session without the need
   for audio transcoding, it is recommended in [I-D.ietf-rtcweb-audio]
   to include in the offer other suitable audio codecs that are
   available to the browser.  This document provides some guidelines on
   the suitable codecs to be considered for WebRTC clients to address
   the most relevant interoperability use cases.

   The codecs considered in this document are recommended to be
   supported and included in the Offer only for WebRTC clients for which
   interoperability with other non WebRTC end points and non WebRTC
   based services is relevant as described in sections 5.1.2, 5.2.2 and
   5.3.2.  Other use cases may justify offering other additional codecs
   to avoid transcodings.  It is the intent of this document to
   inventory and document any other additional interoperability use
   cases and codecs if needed.

2.  Definitions

   Legacy networks: In this draft, legacy networks encompass the
   conversational networks that are already deployed like the PSTN, the
   PLMN, the IMS, H.323 networks.

3.  Rationale for additional WebRTC codecs

   The mandatory implementation of OPUS [RFC6716] in WebRTC clients can
   guarantee the codec interoperability (without transcoding) at the
   state of the art voice quality (better than narrow band "PSTN"
   quality) between WebRTC clients.  The WebRTC technology is however
   expected to be used to communicate with other types of clients using
   other technologies.  It can be used for instance as an access
   technology to 3GPP IMS services (e.g.  VoLTE, ViLTE) or to
   interoperate with fixed or mobile Circuit Switched or VoIP services
   like mobile 3GPP 3G/2G Circuit Switched voice or DECT based VoIP
   telephony.  Consequently, a significant number of calls are likely to
   occur between terminals supporting WebRTC clients and other terminals
   like mobile handsets, fixed VoIP terminals, DECT terminals that do
   not support WebRTC clients nor implement OPUS.  As a consequence,
   these calls are likely to be either of low narrow band PSTN quality
   using G.711 at both ends or affected by transcoding operations.  The
   drawbacks of such transcoding operations are recalled below:





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   o  Degraded user experience with respect to voice quality: voice
      quality is significantly degraded by transcoding.  For instance,
      the degradation is around 0.2 to 0.3 MOS for most of transcoding
      use cases with AMR-WB at 12.65 kbit/s and in the same range for
      other wideband transcoding cases.  It should be stressed that if
      G.711 is used as a fall back codec for interoperation, wideband
      voice quality will be lost.  Such bandwidth reduction effect down
      to narrow band clearly degrades the user perceived quality of
      service leading to shorter and less frequent calls.  Such a switch
      to G.711 is less than desirable or acceptable choice for
      customers.  If transcoding is performed between OPUS and any other
      wideband codec, wideband communication could be maintained but
      with degraded quality (MOS scores of transcoding between AMR-WB
      12.65kbit/s and OPUS at 16 kbit/s in both directions are
      significantly lower than those of AMR-WB at 12.65kbit/s or OPUS at
      16 kbit/s).  Furthermore, in degraded conditions, the addition of
      defects, like audio artifacts due to packet losses, and the audio
      effects resulting from the cascading of different packet loss
      recovery algorithms may result in a quality below the acceptable
      limit for the customers.


   o  Degraded user experience with respect to conversational
      interactivity: the degradation of conversational interactivity is
      due to the increase of end to end latency for both directions that
      is introduced by the transcoding operations.  Transcoding requires
      full de-packetization for decoding of the media stream (including
      mechanisms of de-jitter buffering and packet loss recovery) then
      re-encoding, re-packetization and re-sending.  The delays produced
      by all these operations are additive and may increase the end to
      end delay beyond acceptable limits like with more than 1s end to
      end latency.


   o  Additional costs in networks: transcoding places important
      additional costs on network gateways mainly related to codec
      implementation, codecs license, deployments, testing and
      validation costs.  It must be noted that transcoding of wideband
      to wideband would require more CPU and be more costly than between
      narrowband codecs.


4.  Additional suitable codecs for WebRTC

   The following codecs are considered as relevant suitable codecs with
   respect to the general purpose described in section 4.  This list
   reflects the current status of WebRTC foreseen use cases.  It is not
   limitative and opened to further inclusion of other codecs for which



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   relevant use cases can be identified.  These additional codecs are
   recommended to be included in the offer in addition to OPUS and G.711
   according to the foreseen interoperability cases to be addressed.

4.1.  AMR-WB

4.1.1.  AMR-WB General description

   The Adaptive Multi-Rate WideBand (AMR-WB) is a 3GPP defined speech
   codec that is mandatory to implement in any 3GPP terminal that
   supports wideband speech communication.  It is being used in circuit
   switched mobile telephony services and new multimedia telephony
   services over IP/IMS and 4G/VoLTE, specified by GSMA as voice IMS
   profile for VoLTE in [IR.92].  More detailed information on AMR-WB
   can be found in [IR.36].  [IR.36] includes references for all 3GPP
   AMR-WB related specifications including detailed codec description
   and Source code.

4.1.2.  WebRTC relevant use case for AMR-WB

   The market of voice personal communication is driven by mobile
   terminals.  AMR-WB is now implemented in more than 200 devices models
   and 85 HD mobile networks in 60 countries with a customer base of
   more than 100 million.  A high number of calls are consequently
   likely to occur between WebRTC clients and mobile 3GPP terminals.
   The use of AMR-WB by WebRTC clients would consequently allow
   transcoding free interoperation with all mobile 3GPP wideband
   terminal.  Besides, WebRTC clients running on mobile terminals
   (smartphones) may reuse the AMR-WB codec already implemented on these
   devices.

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

   Guidelines for implementing and using AMR-WB and ensuring
   interoperability with 3GPP mobile services can be found in
   [TS26.114].  In order to ensure interoperability with 4G/VoLTE as
   specified by GSMA, the more specific IMS profile for voice derived
   from [TS26.114] should be considered in [IR.92].  In order to
   maximize the possibility of successful call establishment for WebRTC
   client offering AMR-WB it is important that the WebRTC client:

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

   o  Be capable of operating AMR-WB with any subset of the nine codec
      modes and source controlled rate operation.  Offer at least one



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      AMR-WB configuration with parameter settings as defined in
      Table 6.1 of [TS 26.114].  In order to maximize the
      interoperability and quality this offer does not restrict the
      codec modes offered.  Restrictions in the use of codec modes may
      be included in the answer.

4.2.  AMR

4.2.1.  AMR General description

   Adaptive Multi-Rate (AMR) is a 3GPP defined speech codec that is
   mandatory to implement in any 3GPP terminal that supports voice
   communication, i.e. several hundred millions of terminals.  This
   include both mobile phone calls using GSM and 3G cellular systems as
   well as multimedia telephony services over IP/IMS and 4G/VoLTE, such
   as GSMA voice IMS profile for VoLTE in [IR.92].  In addition to
   impacts listed above, support of AMR can avoid degrading the high
   efficiency over mobile radio access.

4.2.2.  WebRTC relevant use case for AMR

   A user of a WebRTC endpoint on a device integrating an AMR module
   wants to communicate with another user that can only be reached on a
   mobile device that only supports AMR.  Although more and more
   terminal devices are now "HD voice" and support AMR-WB; there is
   still a high number of legacy terminals supporting only AMR
   (terminals with no wideband / HD Voice capabilities) are still used.
   The use of AMR by WebRTC client would consequently allow transcoding
   free interoperation with all mobile 3GPP terminals.  Besides, WebRTC
   client running on mobile terminals (smartphones) may reuse the AMR
   codec already implemented on these devices.

4.2.3.  Guidelines for AMR usage and implementation with WebRTC

   Guidelines for implementing and using AMR with purpose to ensure
   interoperability with 3GPP mobile services can be found in
   [TS26.114].  In order to ensure interoperability with 4G/VoLTE as
   specified by GSMA, the more specific IMS profile for voice derived
   from [TS26.114] should be considered in [IR.92].  In order to
   maximize the possibility of successful call establishment for WebRTC
   client offering AMR, it is important that the WebRTC client:

   o  Be capable of operating AMR with any subset of the eight codec
      modes and source controlled rate operation.

   o  Offer at least one configuration with parameter settings as
      defined in Table 6.1 and Table 6.2 of [TS26.114].  In order to
      maximize the interoperability and quality this offer shall not



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      restrict AMR codec modes offered.  Restrictions in the use of
      codec modes may be included in the answer.

4.3.  G.722

4.3.1.  G.722 General description

   G.722 is an ITU-T defined wideband speech codec.  [G.722] was
   approved by ITU-T in 1988.  It is a royalty free codec that is common
   in a wide range of terminals and end-points supporting wideband
   speech and requiring low complexity.  The complexity of G.722 is
   estimated to 10 MIPS [EN300175-8] which is 2.5 to 3 times lower than
   AMR-WB.  Especially, G.722 has been chosen by ETSI DECT as the
   mandatory wideband codec for New Generation DECT with purpose to
   greatly increase the voice quality by extending the bandwidth from
   narrow band to wideband.  G.722 is the wideband codec required for
   CAT-iq DECT certified terminal and the V2.0 of CAT-iq specifications
   have been approved by GSMA as minimum requirements for HD voice logo
   usage on "fixed" devices; i.e., broadband connections using the G.722
   codec.

4.3.2.  WebRTC relevant use case for G.722

   G.722 is the wideband codec required for DECT CAT-iq terminals.  The
   market for DECT cordeless phones including DECT chipset is more than
   150 Millions per year and CAT-IQ is a registered trade make in 47
   countries worldwide.  G.722 has also been specified by ETSI in
   [TS181005] as mandatory wideband codec for IMS multimedia telephony
   communication service and supplementary services using fixed
   broadband access.  The support of G.722 would consequently allow
   transcoding free IP interoperation between WebRTC client and fixed
   VoIP terminals including DECT / CAT-IQ terminals supporting G.722.
   Besides, WebRTC client running on fixed terminals implementing G.722
   may reuse the G.722 codec already implemented on these devices.

4.3.3.  Guidelines for G.722 usage and implementation

   Guidelines for implementing and using G.722 with purpose to ensure
   interoperability with Multimedia Telephony services overs IMS can be
   found in section 7 of [TS26.114].  Additional information of G.722
   implementation in DECT can be found in [EN300175-8]  and full codec
   description and C source code in [G.722].

4.4.  Other codecs

   Other interoperability use cases may justify the use of other codecs.
   Some further update of this Draft may provide under this section




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   additional use case description and codec implementation guidelines
   for these codecs.

5.  Security Considerations

6.  IANA Considerations

   None.

7.  Acknowledgements

   Thanks to Milan Patel for his review.

8.  References

8.1.  Normative references

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

8.2.  Informative references

   [EN300175-8]
              ETSI, "ETSI EN 300 175-8, v2.5.1: "Digital Enhanced
              Cordless Telecommunications (DECT); Common Interface (CI);
              Part 8: Speech and audio coding and transmission".", 2009.

   [G.722]    ITU, "Recommendation ITU-T G.722 (2012): "7 kHz audio-
              coding within 64 kbit/s".", 2012.

   [I-D.ietf-rtcweb-audio]
              Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
              Requirements", draft-ietf-rtcweb-audio-07 (work in
              progress), October 2014.

   [I-D.ietf-rtcweb-overview]
              Alvestrand, H., "Overview: Real Time Protocols for
              Browser-based Applications", draft-ietf-rtcweb-overview-13
              (work in progress), November 2014.

   [I-D.ietf-rtcweb-use-cases-and-requirements]
              Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
              Time Communication Use-cases and Requirements", draft-
              ietf-rtcweb-use-cases-and-requirements-15 (work in
              progress), December 2014.

   [IR.36]    GSMA, "Adaptive Multirate Wide Band", 2013.




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   [IR.92]    GSMA, "IMS Profile for Voice and SMS", 2013.

   [RFC6716]  Valin, JM., Vos, K., and T. Terriberry, "Definition of the
              Opus Audio Codec", RFC 6716, September 2012.

   [TS181005]
              ETSI, "Telecommunications and Internet converged Services
              and Protocols for Advanced Networking (TISPAN); Service
              and Capability Requirements V3.3.1 (2009-12)", 2009.

   [TS26.114]
              3GPP, "IP Multimedia Subsystem (IMS); Multimedia
              telephony; Media handling and interaction", 2011.

Authors' Addresses

   Stephane Proust
   Orange
   2, avenue Pierre Marzin
   Lannion  22307
   France

   Email: stephane.proust@orange.com


   Espen Berger
   Cisco

   Email: espeberg@cisco.com


   Bernhard Feiten
   Deutsche Telekom

   Email: Bernhard.Feiten@telekom.de


   Bo Burman
   Ericsson

   Email: bo.burman@ericsson.com


   Kalyani Bogineni
   Verizon Wireless

   Email: Kalyani.Bogineni@VerizonWireless.com




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   Miao Lei
   Huawei

   Email: lei.miao@huawei.com


   Enrico Marocco
   Telecom Italia

   Email: enrico.marocco@telecomitalia.it









































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