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rfc:rfc7875

Internet Engineering Task Force (IETF) S. Proust, Ed. Request for Comments: 7875 Orange Category: Informational May 2016 ISSN: 2070-1721

        Additional WebRTC Audio Codecs for Interoperability

Abstract

 To ensure a baseline of interoperability between WebRTC endpoints, a
 minimum set of required codecs is specified.  However, to maximize
 the possibility of establishing the session without the need for
 audio transcoding, it is also recommended to include in the offer
 other suitable audio codecs that are available to the browser.
 This document provides some guidelines on the suitable codecs to be
 considered for WebRTC endpoints to address the use cases most
 relevant to interoperability.

Status of This Memo

 This document is not an Internet Standards Track specification; it is
 published for informational purposes.
 This document is a product of the Internet Engineering Task Force
 (IETF).  It has been approved for publication by the Internet
 Engineering Steering Group (IESG).  Not all documents approved by the
 IESG are a candidate for any level of Internet Standard; see
 Section 2 of RFC 5741.
 Information about the current status of this document, any errata,
 and how to provide feedback on it may be obtained at
 http://www.rfc-editor.org/info/rfc7875.

Proust Informational [Page 1] RFC 7875 WebRTC Audio Codecs for Interop May 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
 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
 carefully, as they describe your rights and restrictions with respect
 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  . . . . . . . . . . . . . . . . . . . . . . . .   3
 2.  Definitions and Abbreviations . . . . . . . . . . . . . . . .   3
 3.  Rationale for Additional WebRTC Codecs  . . . . . . . . . . .   4
 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  . . . . . . . . . . . . . . . . . . . . . . .   6
   4.2.  AMR . . . . . . . . . . . . . . . . . . . . . . . . . . .   6
     4.2.1.  AMR General Description . . . . . . . . . . . . . . .   6
     4.2.2.  WebRTC-Relevant Use Case for AMR  . . . . . . . . . .   7
     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  . . . . . . . . .   8
     4.3.3.  Guidelines for G.722 Usage and Implementation with
             WebRTC  . . . . . . . . . . . . . . . . . . . . . . .   8
 5.  Security Considerations . . . . . . . . . . . . . . . . . . .   8
 6.  References  . . . . . . . . . . . . . . . . . . . . . . . . .   9
   6.1.  Normative References  . . . . . . . . . . . . . . . . . .   9
   6.2.  Informative References  . . . . . . . . . . . . . . . . .  10
 Acknowledgements  . . . . . . . . . . . . . . . . . . . . . . . .  12
 Contributors  . . . . . . . . . . . . . . . . . . . . . . . . . .  12
 Author's Address  . . . . . . . . . . . . . . . . . . . . . . . .  12

Proust Informational [Page 2] RFC 7875 WebRTC Audio Codecs for Interop May 2016

1. Introduction

 As indicated in [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 of establishing the session without the need for audio
 transcoding, it is recommended in [RFC7874] to include in the offer
 other suitable audio codecs beyond those that are mandatory to
 implement.  This document provides some guidelines on the suitable
 codecs to be considered for WebRTC endpoints to address the use cases
 most relevant to interoperability.
 The codecs considered in this document are recommended to be
 supported and included in the offer, only for WebRTC endpoints for
 which interoperability with other non-WebRTC endpoints and non-
 WebRTC-based services is relevant as described in Sections 4.1.2,
 4.2.2, and 4.3.2.  Other use cases may justify offering other
 additional codecs to avoid transcoding.

2. Definitions 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 (VoLTE) radio access [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 [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 Subsystems
 o  LTE: Long Term Evolution (3GPP "4G" wireless data transmission
    standard)
 o  MOS: Mean Opinion Score, defined in ITU-T Recommendation P.800
    [P.800]

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 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
 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, and 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 the
    transcoding use cases with AMR-WB codec (Section 4.1) at 12.65
    kbit/s and in the same range for other wideband transcoding cases.
    It should be stressed that if G.711 is used as a fallback 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 a choice for customers.
    If transcoding is performed between Opus and any other wideband
    codec, wideband communication could be maintained but with
    degraded quality (MOSs of transcoding between AMR-WB at 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
    (a) audio artifacts due to packet losses and (b) audio effects due
    to the cascading of different packet loss recovery algorithms, may
    result in a quality below the acceptable limit for the customers.

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

4. Additional Suitable Codecs for WebRTC

 The following are considered relevant codecs with respect to the
 general purpose described in Section 3.  This list reflects the
 current status of foreseen use cases for WebRTC.  It is not
 limitative; it is open to inclusion of other codecs for which
 relevant use cases can be identified.  It's recommended to include
 codecs (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 specially used for voice over LTE as
 specified by GSMA in [IR.92].  More detailed information on AMR-WB
 can be found in [IR.36].  References for AMR-WB-related
 specifications including the detailed codec description and source
 code are in [TS26.171], [TS26.173], [TS26.190], and [TS26.204].

4.1.2. WebRTC-Relevant Use Case 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", where "HD" stands for "High
 Definition".  Consequently, a high number of calls are likely to
 occur between WebRTC endpoints and mobile 3GPP terminals offering

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 AMR-WB.  Thus, the use of AMR-WB by WebRTC endpoints would 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 those
 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 a bandwidth-efficient format and one speech frame encapsulated
 in each RTP packet.  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
 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 the preferred payload type with respect
    to other narrow-band codecs (AMR, G.711, etc.) and with a
    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.
 o  Offer at least one AMR-WB configuration with parameter settings as
    defined in Table 6.1 of [TS26.114].  In order to maximize
    interoperability and quality, this offer does not restrict the
    codec modes offered.  Restrictions on 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 includes both mobile phone calls using GSM and
 3G cellular systems as well as multimedia telephony services over IP/
 IMS and 4G/VoLTE, such as the GSMA voice IMS profile for VoLTE in
 [IR.92].  References for AMR-related specifications including
 detailed codec description and source code are [TS26.071],
 [TS26.073], [TS26.090], and [TS26.104].

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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.

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 packet.  Further guidelines for implementing and using AMR with
 purpose to ensure interoperability with 3GPP mobile services can be
 found in [TS26.114].  In order to ensure interoperability with 4G/
 VoLTE as specified by GSMA, the more specific IMS profile for voice
 derived from [TS26.114] should be considered in [IR.92].  In order to
 maximize the possibility of successful call establishment for WebRTC
 endpoints offering AMR, it is important that the WebRTC endpoints:
 o  Be capable of operating AMR with any subset of the eight codec
    modes and source-controlled rate operation.
 o  Offer at least one configuration with parameter settings as
    defined in Tables 6.1 and 6.2 of [TS26.114].  In order to maximize
    the interoperability and quality, this offer shall not restrict
    AMR codec modes offered.  Restrictions on 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 the 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.  In particular, G.722 has been chosen by ETSI DECT as the
 mandatory wideband codec for New Generation DECT in order to greatly
 increase the voice quality by extending the bandwidth from narrow
 band to wideband.  G.722 is the wideband codec required for terminals
 that are certified as CAT-iq DECT, and version 2.0 of the CAT-iq

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 specifications have been approved by GSMA as the minimum requirements
 for the "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
 cordless 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 the mandatory wideband codec for IMS
 multimedia telephony communication service and supplementary services
 using fixed broadband access.  The support of G.722 would
 consequently allow transcoding-free IP interoperation between WebRTC
 endpoints and fixed VoIP terminals including DECT CAT-iq terminals
 supporting G.722.  Besides, WebRTC endpoints running on fixed
 terminals that implement G.722 may reuse the G.722 codec already
 implemented on these devices.

4.3.3. Guidelines for G.722 Usage and Implementation with WebRTC

 The payload format to be used for G.722 is defined in [RFC3551] with
 each octet of the stream of octets produced by the codec to be octet-
 aligned in an RTP packet.  The sampling frequency for the G.722 codec
 is 16 kHz, but the RTP clock rate is set to 8000 Hz in SDP to stay
 backward compatible with an erroneous definition in the original
 version of the RTP audio/video profile.  Further guidelines for
 implementing and using G.722 to ensure interoperability with
 multimedia telephony services over IMS can be found in Section 7 of
 [TS26.114].  Additional information about the G.722 implementation in
 DECT can be found in [EN300175-8], and the full codec description and
 C source code are in [G.722].

5. Security Considerations

 Relevant security considerations can be found in [RFC7874], "WebRTC
 Audio Codec and Processing Requirements".  Implementers making use of
 the additional codecs considered in this document are advised to also
 refer more specifically to the "Security Considerations" sections of
 [RFC4867] (for AMR and AMR-WB) and [RFC3551] (for the RTP audio/video
 profile).

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6. References

6.1. Normative References

 [G.722]    ITU-T, "7 kHz audio-coding within 64 kbit/s", ITU-T
            Recommendation G.722, September 2012,
            <http://www.itu.int/rec/T-REC-G.722-201209-I/en>.
 [IR.92]    GSMA, "IMS Profile for Voice and SMS", IR.92, Version 9.0,
            April 2015, <http://www.gsma.com/newsroom/all-documents/
            ir-92-ims-profile-for-voice-and-sms/>.
 [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
            Video Conferences with Minimal Control", STD 65, RFC 3551,
            DOI 10.17487/RFC3551, July 2003,
            <http://www.rfc-editor.org/info/rfc3551>.
 [RFC4867]  Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q. Xie,
            "RTP Payload Format and File Storage Format for the
            Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband
            (AMR-WB) Audio Codecs", RFC 4867, DOI 10.17487/RFC4867,
            April 2007, <http://www.rfc-editor.org/info/rfc4867>.
 [RFC7874]  Valin, JM. and C. Bran, "WebRTC Audio Codec and Processing
            Requirements", RFC 7874, DOI 10.17487/RFC7874, May 2016,
            <http://www.rfc-editor.org/info/rfc7874>.
 [TS26.071] 3GPP, "Mandatory Speech Codec speech processing functions;
            AMR Speech CODEC; General description", 3GPP TS 26.171
            v13.0.0, December 2015,
            <http://www.3gpp.org/DynaReport/26071.htm>.
 [TS26.073] 3GPP, "ANSI C code for the Adaptive Multi Rate (AMR)
            speech codec", 3GPP TS 26.073 v13.0.0, December 2015,
            <http://www.3gpp.org/DynaReport/26073.htm>.
 [TS26.090] 3GPP, "Mandatory Speech Codec speech processing functions;
            Adaptive Multi-Rate (AMR) speech codec; Transcoding
            functions.", 3GPP TS 26.090 v13.0.0, December 2015,
            <http://www.3gpp.org/DynaReport/26090.htm>.
 [TS26.104] 3GPP, "ANSI C code for the floating-point Adaptive Multi
            Rate (AMR) speech codec.", 3GPP TS 26.104 v13.0.0,
            December 2015, <http://www.3gpp.org/DynaReport/26090.htm>.

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 [TS26.114] 3GPP, "IP Multimedia Subsystem (IMS); Multimedia
            telephony; Media handling and interaction", 3GPP TS 26.114
            v13.3.0, March 2016,
            <http://www.3gpp.org/DynaReport/26114.htm>.
 [TS26.171] 3GPP, "Speech codec speech processing functions; Adaptive
            Multi-Rate - Wideband (AMR-WB) speech codec; General
            description.", 3GPP TS 26.171 v13.0.0, December 2015,
            <http://www.3gpp.org/DynaReport/26171.htm>.
 [TS26.173] 3GPP, "ANSI-C code for the Adaptive Multi-Rate - Wideband
            (AMR-WB) speech codec", 3GPP TS 26.173 v13.1.0, March
            2016, <http://www.3gpp.org/DynaReport/26173.htm>.
 [TS26.190] 3GPP, "Speech codec speech processing functions; Adaptive
            Multi-Rate - Wideband (AMR-WB) speech codec; Transcoding
            functions", 3GPP TS 26.190 v13.0.0, December 2015,
            <http://www.3gpp.org/DynaReport/26190.htm>.
 [TS26.204] 3GPP, "Speech codec speech processing functions; Adaptive
            Multi-Rate - Wideband (AMR-WB) speech codec; ANSI-C
            code.", 3GPP TS 26.204 v13.1.0, March 2016,
            <http://www.3gpp.org/DynaReport/26204.htm>.

6.2. Informative References

 [EN300175-1]
            ETSI, "Digital Enhanced Cordless Telecommunications
            (DECT); Common Interface (CI); Part 1: Overview", ETSI
            EN 300 175-1, v2.6.1, 2015,
            <http://www.etsi.org/deliver/etsi_en/300100_300199/
            30017501/02.06.01_60/en_30017501v020601p.pdf>.
 [EN300175-8]
            ETSI, "Digital Enhanced Cordless Telecommunications
            (DECT); Common Interface (CI); Part 8: Speech and audio
            coding and transmission.", ETSI EN 300 175-8, v2.6.1,
            2015,
            <http://www.etsi.org/deliver/etsi_en/300100_300199/
            30017508/02.06.01_60/en_30017508v020601p.pdf>.
 [G.711]    ITU-T, "Pulse code modulation (PCM) of voice frequencies",
            ITU-T Recommendation G.711, November 1988,
            <http://www.itu.int/rec/T-REC-G.711-198811-I/en>.

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 [IR.36]    GSMA, "Adaptive Multirate Wide Band", IR.36, Version 3.0,
            September 2014,
            <http://www.gsma.com/newsroom/all-documents/
            official-document-ir-36-adaptive-multirate-wide-band>.
 [OVERVIEW] Alvestrand, H., "Overview: Real Time Protocols for
            Browser-based Applications", Work in Progress,
            draft-ietf-rtcweb-overview-15, January 2016.
 [P.800]    ITU-T, "Methods for subjective determination of
            transmission quality", ITU-T Recommendation P.800, August
            1996, <https://www.itu.int/rec/T-REC-P.800-199608-I/en>.
 [RFC6716]  Valin, JM., Vos, K., and T. Terriberry, "Definition of the
            Opus Audio Codec", RFC 6716, DOI 10.17487/RFC6716,
            September 2012, <http://www.rfc-editor.org/info/rfc6716>.
 [TS181005] ETSI, "Telecommunications and Internet converged Services
            and Protocols for Advanced Networking (TISPAN); Service
            and Capability Requirements V3.3.1 (2009-12)", ETSI
            TS 181005, 2009,
            <http://www.etsi.org/deliver/etsi_ts/181000_181099/
            181005/03.03.01_60/ts_181005v030301p.pdf>.
 [TS23.002] 3GPP, "Network architecture", 3GPP TS23.002 v13.5.0, March
            2016, <http://www.3gpp.org/dynareport/23002.htm>.

Proust Informational [Page 11] RFC 7875 WebRTC Audio Codecs for Interop May 2016

Acknowledgements

 We would like to thank Magnus Westerlund, Barry Dingle, and Sanjay
 Mishra who carefully reviewed the document and helped to improve it.

Contributors

 The following individuals contributed significantly to this document:
 o  Stephane Proust, Orange, stephane.proust@orange.com
 o  Espen Berger, Cisco, espeberg@cisco.com
 o  Bernhard Feiten, Deutsche Telekom, Bernhard.Feiten@telekom.de
 o  Bo Burman, Ericsson, bo.burman@ericsson.com
 o  Kalyani Bogineni, Verizon Wireless,
    Kalyani.Bogineni@VerizonWireless.com
 o  Mia Lei, Huawei, lei.miao@huawei.com
 o  Enrico Marocco, Telecom Italia, enrico.marocco@telecomitalia.it
 though only the editor is listed on the front page.

Author's Address

 Stephane Proust (editor)
 Orange
 2, avenue Pierre Marzin
 Lannion  22307
 France
 Email: stephane.proust@orange.com

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