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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, Ed.
Internet-Draft                                                    Orange
Intended status: Informational                         February 10, 2016
Expires: August 13, 2016


          Additional WebRTC audio codecs for interoperability.
             draft-ietf-rtcweb-audio-codecs-for-interop-05

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 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 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
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   Internet-Drafts are draft documents valid for a maximum of six months
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   This Internet-Draft will expire on August 13, 2016.

Copyright Notice

   Copyright (c) 2016 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
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   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect



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   to this document.  Code Components extracted from this document must
   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  . . . . . . . . . . . . . . . . . . . . . . . .   2
   2.  Definition and abbreviations  . . . . . . . . . . . . . . . .   3
   3.  Rationale for additional WebRTC codecs  . . . . . . . . . . .   3
   4.  Additional suitable codecs for WebRTC . . . . . . . . . . . .   5
     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  . . . . . . . . . . . . . . . . . . . . . . .   7
     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 . . . .   8
   5.  Security Considerations . . . . . . . . . . . . . . . . . . .   8
   6.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   8
   7.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .   8
   8.  References  . . . . . . . . . . . . . . . . . . . . . . . . .   9
     8.1.  Normative references  . . . . . . . . . . . . . . . . . .   9
     8.2.  Informative references  . . . . . . . . . . . . . . . . .  10
   Author's Address  . . . . . . . . . . . . . . . . . . . . . . . .  11

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 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 most relevant interoperability use cases.

   The codecs considered in this document are recommended to be
   supported and included in the Offer only for WebRTC endpoints for



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   which interoperability with other non-WebRTC endpoints and non-WebRTC
   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
   other additional codecs to avoid transcoding.

2.  Definition and 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 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-WB: Adaptive Multi-Rate WideBand.

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

   o  DECT: Digital Enhanced Cordless Telecommunications

   o  IMS: IP Multimedia Subsystem

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

   o  MOS: Mean Opinion Score

   o  PSTN:Public Switched Telephone Network

   o  PLMN: Public Land Mobile Network

   o  VoLTE: Voice Over LTE

3.  Rationale for additional WebRTC codecs

   The mandatory implementation of OPUS [RFC6716] in WebRTC endpoints
   can guarantee codec interoperability (without transcoding) at state
   of the art voice quality (better than 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 or VoIP services



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   like mobile Circuit Switched voice over 3GPP 2G/3G mobile networks
   [TS23.002] or 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, DECT terminals that do not support WebRTC
   endpoints nor implement OPUS.  As a consequence, these calls are
   likely to be either of low 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 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 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 12.65 kbit/s and OPUS at 16 kbit/s in
      both directions are significantly lower than those of AMR-WB at
      12.65 kbit/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 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




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      wideband would require more CPU processing and be more costly than
      transcoding between narrowband codecs.


4.  Additional 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.  It is not
   limitative and opened to further inclusion of other codecs for which
   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.  It is especially 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
   specifications including detailed codec description and source code
   are in [TS26.171], [TS26.173], [TS26.190], [TS26.204].

4.1.2.  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 high number of calls are consequently
   likely to occur between WebRTC endpoints and mobile 3GPP terminals
   offering AMR-WB.  The use of AMR-WB by WebRTC endpoints would
   consequently allow 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 devices.

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]
   with bandwidth efficient format and one speech frame encapsulated in
   each RTP packets.  Further 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



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   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 it is important that the WebRTC endpoints:

   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
      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
      AMR-WB configuration with parameter settings as defined in
      Table 6.1 of [TS26.114].  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.  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.References for
   AMR related specifications including detailed codec description and
   source code are in [TS26.071], [TS26.073], [TS26.090], [TS26.104].

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 are
   still a high number of legacy terminals supporting only AMR
   (terminals with no wideband / HD 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.








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4.2.3.  Guidelines for AMR usage and 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 packets.  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 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
      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 [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 endpoints 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 terminals 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.  DECT
   cordeless phones are still widely used to offer short range wireless
   connection to PSTN or VoIP services.  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 endpoints and



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   fixed VoIP terminals including DECT / CAT-IQ terminals supporting
   G.722.  Besides, WebRTC endpoints 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

   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 G.722 is 16 kHz
   but the rtp clock rate is set to 8000Hz in SDP to stay backward
   compatible with an erroneous definition in the original version of
   the RTP A/V 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 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].

5.  Security Considerations

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



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   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 the document and helped to
   improve it.

8.  References

8.1.  Normative references

   [G.722]    ITU, "Recommendation ITU-T G.722 (2012): 7 kHz audio-
              coding within 64 kbit/s", 2012-09.

   [I-D.ietf-rtcweb-audio]
              Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
              Requirements", draft-ietf-rtcweb-audio-09 (work in
              progress), November 2015.

   [IR.92]    GSMA, "IMS Profile for Voice and SMS V9.0", April 2015.

   [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>.

   [TS26.071]
              3GPP, "3GPP TS 26.071 v12.0.0: Recommendation ITU-T G.722
              (2012): "Mandatory Speech Codec speech processing
              functions; AMR Speech CODEC; General description".",
              2014-09.

   [TS26.073]
              3GPP, "3GPP TS 26.073 v12.0.0: ANSI C code for the
              Adaptive Multi Rate (AMR) speech codec", 2014-09.

   [TS26.090]
              3GPP, "3GPP TS 26.090 v12.0.0: Mandatory Speech Codec
              speech processing functions; Adaptive Multi-Rate (AMR)
              speech codec; Transcoding functions.", 2014-09.




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   [TS26.104]
              3GPP, "3GPP TS 26.104 v12.0.0: ANSI C code for the
              floating-point Adaptive Multi Rate (AMR) speech codec.",
              2014-09.

   [TS26.114]
              3GPP, "IP Multimedia Subsystem (IMS); Multimedia
              telephony; Media handling and interaction V13.0.0", June
              2015.

   [TS26.171]
              3GPP, "3GPP TS 26.071 v12.0.0: Recommendation ITU-T G.722
              (2012): "Speech codec speech processing functions;
              Adaptive Multi-Rate - Wideband (AMR-WB) speech codec;
              General description".", 2014-09.

   [TS26.173]
              3GPP, "3GPP TS 26.073 v12.1.0: ANSI-C code for the
              Adaptive Multi-Rate - Wideband (AMR-WB) speech codec.",
              2015-03.

   [TS26.190]
              3GPP, "3GPP TS 26.090 v12.0.0: Speech codec speech
              processing functions; Adaptive Multi-Rate - Wideband (AMR-
              WB) speech codec; Transcoding functions.", 2014-09.

   [TS26.204]
              3GPP, "3GPP TS 26.104 v12.1.0: Speech codec speech
              processing functions; Adaptive Multi-Rate - Wideband (AMR-
              WB) speech codec; ANSI-C code.", 2015-03.

8.2.  Informative references

   [EN300175-1]
              ETSI, "ETSI EN 300 175-1, Digital Enhanced Cordless
              Telecommunications (DECT); Common Interface (CI); Part 1:
              Overview v2.5.1", 2009.

   [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.711]    ITU, "Recommendation ITU-T G.711 (2012): Pulse code
              modulation (PCM) of voice frequencies", 1988-11.






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

   [IR.36]    GSMA, "Adaptive Multirate Wide Band V3.0", September 2014.

   [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)", 2009.

   [TS23.002]
              3GPP, "3GPP TS 23.002 v13.3.0: Network architecture",
              2015-09.

Author's Address

   Stephane Proust (editor)
   Orange
   2, avenue Pierre Marzin
   Lannion  22307
   France

   Email: stephane.proust@orange.com






















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