





Network Working Group                                        Q. Xie, Ed.
Request for Comments: 3557                                Motorola, Inc.
Category: Standards Track                                      July 2003


                         RTP Payload Format for
European Telecommunications Standards Institute (ETSI) European Standard
           ES 201 108 Distributed Speech Recognition Encoding

Status of this Memo

   This document specifies an Internet standards track protocol for the
   Internet community, and requests discussion and suggestions for
   improvements.  Please refer to the current edition of the "Internet
   Official Protocol Standards" (STD 1) for the standardization state
   and status of this protocol.  Distribution of this memo is unlimited.

Copyright Notice

   Copyright (C) The Internet Society (2003).  All Rights Reserved.

Abstract

   This document specifies an RTP payload format for encapsulating
   European Telecommunications Standards Institute (ETSI) European
   Standard (ES) 201 108 front-end signal processing feature streams for
   distributed speech recognition (DSR) systems.
























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Table of Contents

   1.  Conventions and Acronyms . . . . . . . . . . . . . . . . . . .  2
   2.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  2
       2.1.  ETSI ES 201 108 DSR Front-end Codec. . . . . . . . . . .  3
       2.2.  Typical Scenarios for Using DSR Payload Format . . . . .  4
   3.  ES 201 108 DSR RTP Payload Format. . . . . . . . . . . . . . .  5
       3.1.  Consideration on Number of FPs in Each RTP Packet. . . .  6
       3.2.  Support for Discontinuous Transmission . . . . . . . . .  6
   4.  Frame Pair Formats . . . . . . . . . . . . . . . . . . . . . .  7
       4.1.  Format of Speech and Non-speech FPs. . . . . . . . . . .  7
       4.2.  Format of Null FP. . . . . . . . . . . . . . . . . . . .  8
       4.3.  RTP header usage . . . . . . . . . . . . . . . . . . . .  8
   5.  IANA Considerations. . . . . . . . . . . . . . . . . . . . . .  9
       5.1.  Mapping MIME Parameters into SDP . . . . . . . . . . . . 10
   6.  Security Considerations. . . . . . . . . . . . . . . . . . . . 11
   7.  Contributors . . . . . . . . . . . . . . . . . . . . . . . . . 11
   8.  Acknowledgments. . . . . . . . . . . . . . . . . . . . . . . . 11
   9.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 11
       9.1.  Normative References . . . . . . . . . . . . . . . . . . 11
       9.2.  Informative References . . . . . . . . . . . . . . . . . 12
   10. IPR Notices. . . . . . . . . . . . . . . . . . . . . . . . . . 12
   11. Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . 13
   12. Editor's Address . . . . . . . . . . . . . . . . . . . . . . . 14
   13. Full Copyright Statement . . . . . . . . . . . . . . . . . . . 15

1.  Conventions and Acronyms

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in [RFC2119].

   The following acronyms are used in this document:

   DSR  - Distributed Speech Recognition

   ETSI - the European Telecommunications Standards Institute

   FP   - Frame Pair

   DTX  - Discontinuous Transmission

2.  Introduction

   Motivated by technology advances in the field of speech recognition,
   voice interfaces to services (such as airline information systems,
   unified messaging) are becoming more prevalent.  In parallel, the
   popularity of mobile devices has also increased dramatically.



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   However, the voice codecs typically employed in mobile devices were
   designed to optimize audible voice quality and not speech recognition
   accuracy, and using these codecs with speech recognizers can result
   in poor recognition performance.  For systems that can be accessed
   from heterogeneous networks using multiple speech codecs, recognition
   system designers are further challenged to accommodate the
   characteristics of these differences in a robust manner.  Channel
   errors and lost data packets in these networks result in further
   degradation of the speech signal.

   In traditional systems as described above, the entire speech
   recognizer lies on the server.  It is forced to use incoming speech
   in whatever condition it arrives after the network decodes the
   vocoded speech.  To address this problem, we use a distributed speech
   recognition (DSR) architecture.  In such a system, the remote device
   acts as a thin client, also known as the front-end, in communication
   with a speech recognition server, also called a speech engine.  The
   remote device processes the speech, compresses the data, and adds
   error protection to the bitstream in a manner optimal for speech
   recognition.  The speech engine then uses this representation
   directly, minimizing the signal processing necessary and benefiting
   from enhanced error concealment.

   To achieve interoperability with different client devices and speech
   engines, a common format is needed.  Within the "Aurora" DSR working
   group of the European Telecommunications Standards Institute (ETSI),
   a payload has been defined and was published as a standard [ES201108]
   in February 2000.

   For voice dialogues between a caller and a voice service, low latency
   is a high priority along with accurate speech recognition.  While
   jitter in the speech recognizer input is not particularly important,
   many issues related to speech interaction over an IP-based connection
   are still relevant.  Therefore, it is desirable to use the DSR
   payload in an RTP-based session.

2.1  ETSI ES 201 108 DSR Front-end Codec

   The ETSI Standard ES 201 108 for DSR [ES201108] defines a signal
   processing front-end and compression scheme for speech input to a
   speech recognition system.  Some relevant characteristics of this
   ETSI DSR front-end codec are summarized below.

   The coding algorithm, a standard mel-cepstral technique common to
   many speech recognition systems, supports three raw sampling rates: 8
   kHz, 11 kHz, and 16 kHz.  The mel-cepstral calculation is a frame-
   based scheme that produces an output vector every 10 ms.




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   After calculation of the mel-cepstral representation, the
   representation is first quantized via split-vector quantization to
   reduce the data rate of the encoded stream.  Then, the quantized
   vectors from two consecutive frames are put into an FP, as described
   in more detail in Section 4.1.

2.2  Typical Scenarios for Using DSR Payload Format

   The diagrams in Figure 1 show some typical use scenarios of the ES
   201 108 DSR RTP payload format.

   +--------+                     +----------+
   |IP USER |  IP/UDP/RTP/DSR     |IP SPEECH |
   |TERMINAL|-------------------->|  ENGINE  |
   |        |                     |          |
   +--------+                     +----------+

     a) IP user terminal to IP speech engine

   +--------+  DSR over      +-------+                +----------+
   | Non-IP |  Circuit link  |       | IP/UDP/RTP/DSR |IP SPEECH |
   |  USER  |:::::::::::::::>|GATEWAY|--------------->|  ENGINE  |
   |TERMINAL|  ETSI payload  |       |                |          |
   +--------+  format        +-------+                +----------+

     b) non-IP user terminal to IP speech engine via a gateway

   +--------+                  +-------+  DSR over       +----------+
   |IP USER |  IP/UDP/RTP/DSR  |       |  circuit link   |  Non-IP  |
   |TERMINAL|----------------->|GATEWAY|::::::::::::::::>|  SPEECH  |
   |        |                  |       |  ETSI payload   |  ENGINE  |
   +--------+                  +-------+  format         +----------+

     c) IP user terminal to non-IP speech engine via a gateway

         Figure 1: Typical Scenarios for Using DSR Payload Format.

   For the different scenarios in Figure 1, the speech recognizer always
   resides in the speech engine.  A DSR front-end encoder inside the
   User Terminal performs front-end speech processing and sends the
   resultant data to the speech engine in the form of "frame pairs"
   (FPs).  Each FP contains two sets of encoded speech vectors
   representing 20ms of original speech.








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3.  ES 201 108 DSR RTP Payload Format

   An ES 201 108 DSR RTP payload datagram consists of a standard RTP
   header [RFC3550] followed by a DSR payload.  The DSR payload itself
   is formed by concatenating a series of ES 201 108 DSR FPs (defined in
   Section 4).

   FPs are always packed bit-contiguously into the payload octets
   beginning with the most significant bit.  For ES 201 108 front-end,
   the size of each FP is 96 bits or 12 octets (see Sections 4.1 and
   4.2).  This ensures that a DSR payload will always end on an octet
   boundary.

   The following example shows a DSR RTP datagram carrying a DSR payload
   containing three 96-bit-long FPs (bit 0 is the MSB):

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   \                                                               \
   /                    RTP header in [RFC3550]                    /
   \                                                               \
   +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
   |                                                               |
   +                                                               +
   |                         FP #1 (96 bits)                       |
   +                                                               +
   |                                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                                               |
   +                                                               +
   |                         FP #2 (96 bits)                       |
   +                                                               +
   |                                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                                               |
   +                                                               +
   |                         FP #3 (96 bits)                       |
   +                                                               +
   |                                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

          Figure 2. An example of an ES 201 108 DSR RTP payload.








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3.1 Consideration on Number of FPs in Each RTP Packet

   The number of FPs per payload packet should be determined by the
   latency and bandwidth requirements of the DSR application using this
   payload format.  In particular, using a smaller number of FPs per
   payload packet in a session will result in lowered bandwidth
   efficiency due to the RTP/UDP/IP header overhead, while using a
   larger number of FPs per packet will cause longer end-to-end delay
   and hence increased recognition latency.  Furthermore, carrying a
   larger number of FPs per packet will increase the possibility of
   catastrophic packet loss; the loss of a large number of consecutive
   FPs is a situation most speech recognizers have difficulty dealing
   with.

   It is therefore RECOMMENDED that the number of FPs per DSR payload
   packet be minimized, subject to meeting the application's
   requirements on network bandwidth efficiency.  RTP header compression
   techniques, such as those defined in [RFC2508] and [RFC3095], should
   be considered to improve network bandwidth efficiency.

3.2 Support for Discontinuous Transmission

   The DSR RTP payloads may be used to support discontinuous
   transmission (DTX) of speech, which allows that DSR FPs are sent only
   when speech has been detected at the terminal equipment.

   In DTX a set of DSR frames coding an unbroken speech segment
   transmitted from the terminal to the server is called a transmission
   segment.  A DSR frame inside such a transmission segment can be
   either a speech frame or a non-speech frame, depending on the nature
   of the section of the speech signal it represents.

   The end of a transmission segment is determined at the sending end
   equipment when the number of consecutive non-speech frames exceeds a
   pre-set threshold, called the hangover time.  A typical value used
   for the hangover time is 1.5 seconds.

   After all FPs in a transmission segment are sent, the front-end
   SHOULD indicate the end of the current transmission segment by
   sending one or more Null FPs (defined in Section 4.2).











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4.  Frame Pair Formats

4.1 Format of Speech and Non-speech FPs

   The following mel-cepstral frame MUST be used, as defined in
   [ES201108]:

   As defined in [ES201108], pairs of the quantized 10ms mel-cepstral
   frames MUST be grouped together and protected with a 4-bit CRC,
   forming a 92-bit long FP:

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                      Frame #1  (44 bits)                      |
   +                       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                       |          Frame #2 (44 bits)           |
   +-+-+-+-+-+-+-+-+-+-+-+-+                       +-+-+-+-+-+-+-+-+
   |                                               | CRC   |0|0|0|0|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   The length of each frame is 44 bits representing 10ms of voice. The
   following mel-cepstral frame formats MUST be used when forming an FP:

   Frame #1 in FP:
   ===============
       (MSB)                                     (LSB)
         0     1     2     3     4     5     6     7
      +-----+-----+-----+-----+-----+-----+-----+-----+
      :  idx(2,3) |            idx(0,1)               |    Octet 1
      +-----+-----+-----+-----+-----+-----+-----+-----+
      :       idx(4,5)        |     idx(2,3) (cont)   :    Octet 2
      +-----+-----+-----+-----+-----+-----+-----+-----+
      |             idx(6,7)              |idx(4,5)(cont)  Octet 3
      +-----+-----+-----+-----+-----+-----+-----+-----+
       idx(10,11) |              idx(8,9)             |    Octet 4
      +-----+-----+-----+-----+-----+-----+-----+-----+
      :       idx(12,13)      |   idx(10,11) (cont)   :    Octet 5
      +-----+-----+-----+-----+-----+-----+-----+-----+
                              |   idx(12,13) (cont)   :    Octet 6/1
                              +-----+-----+-----+-----+










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   Frame #2 in FP:
   ===============
       (MSB)                                     (LSB)
         0     1     2     3     4     5     6     7
      +-----+-----+-----+-----+
      :        idx(0,1)       |                            Octet 6/2
      +-----+-----+-----+-----+-----+-----+-----+-----+
      |              idx(2,3)             |idx(0,1)(cont)  Octet 7
      +-----+-----+-----+-----+-----+-----+-----+-----+
      :  idx(6,7) |              idx(4,5)             |    Octet 8
      +-----+-----+-----+-----+-----+-----+-----+-----+
      :        idx(8,9)       |      idx(6,7) (cont)  :    Octet 9
      +-----+-----+-----+-----+-----+-----+-----+-----+
      |          idx(10,11)               |idx(8,9)(cont)  Octet 10
      +-----+-----+-----+-----+-----+-----+-----+-----+
      |                   idx(12,13)                  |    Octet 11
      +-----+-----+-----+-----+-----+-----+-----+-----+

   Therefore, each FP represents 20ms of original speech.  Note, as
   shown above, each FP MUST be padded with 4 zeros to the end in order
   to make it aligned to the 32-bit word boundary.  This makes the size
   of an FP 96 bits, or 12 octets.  Note, this padding is separate from
   padding indicated by the P bit in the RTP header.

   The 4-bit CRC MUST be calculated using the formula defined in 6.2.4
   in [ES201108]. The definition of the indices and the determination of
   their value are also described in [ES201108].

4.2 Format of Null FP

   A Null FP for the ES 201 108 front-end codec is defined by setting
   the content of the first and second frame in the FP to null (i.e.,
   filling the first 88 bits of the FP with 0's).  The 4-bit CRC MUST be
   calculated the same way as described in 6.2.4 in [ES201108], and 4
   zeros MUST be padded to the end of the Null FP to make it 32-bit word
   aligned.

4.3 RTP header usage

   The format of the RTP header is specified in [RFC3550].  This payload
   format uses the fields of the header in a manner consistent with that
   specification.

   The RTP timestamp corresponds to the sampling instant of the first
   sample encoded for the first FP in the packet.  The timestamp clock
   frequency is the same as the sampling frequency, so the timestamp
   unit is in samples.




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   As defined by ES 201 108 front-end codec, the duration of one FP is
   20 ms, corresponding to 160, 220, or 320 encoded samples with
   sampling rate of 8, 11, or 16 kHz being used at the front-end,
   respectively. Thus, the timestamp is increased by 160, 220, or 320
   for each consecutive FP, respectively.

   The DSR payload for ES 201 108 front-end codes is always an integral
   number of octets.  If additional padding is required for some other
   purpose, then the P bit in the RTP in the header may be set and
   padding appended as specified in [RFC3550].

   The RTP header marker bit (M) should be set following the general
   rules defined in [RFC3551].

   The assignment of an RTP payload type for this new packet format is
   outside the scope of this document, and will not be specified here.
   It is expected that the RTP profile under which this payload format
   is being used will assign a payload type for this encoding or specify
   that the payload type is to be bound dynamically.

5.  IANA Considerations

   One new MIME subtype registration is required for this payload type,
   as defined below.

   This section also defines the optional parameters that may be used to
   describe a DSR session.  The parameters are defined here as part of
   the MIME subtype registration.  A mapping of the parameters into the
   Session Description Protocol (SDP) [RFC2327] is also provided in 5.1
   for those applications that use SDP.

   Media Type name: audio

   Media subtype name: dsr-es201108

   Required parameters: none

   Optional parameters:

   rate: Indicates the sample rate of the speech.  Valid values include:
      8000, 11000, and 16000.  If this parameter is not present, 8000
      sample rate is assumed.

   maxptime: The maximum amount of media which can be encapsulated in
      each packet, expressed as time in milliseconds.  The time shall be
      calculated as the sum of the time the media present in the packet
      represents.  The time SHOULD be a multiple of the frame pair size
      (i.e., one FP <-> 20ms).



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      If this parameter is not present, maxptime is assumed to be 80ms.

      Note, since the performance of most speech recognizers are
      extremely sensitive to consecutive FP losses, if the user of the
      payload format expects a high packet loss ratio for the session,
      it MAY consider to explicitly choose a maxptime value for the
      session that is shorter than the default value.

   ptime: see RFC2327 [RFC2327].

   Encoding considerations : This type is defined for transfer via RTP
      [RFC3550] as described in Sections 3 and 4 of RFC 3557.

   Security considerations : See Section 6 of RFC 3557.

   Person & email address to contact for further information:
      Qiaobing.Xie@motorola.com

   Intended usage: COMMON.  It is expected that many VoIP applications
      (as well as mobile applications) will use this type.

   Author/Change controller:
      Qiaobing.Xie@motorola.com
      IETF Audio/Video transport working group

5.1 Mapping MIME Parameters into SDP

   The information carried in the MIME media type specification has a
   specific mapping to fields in the Session Description Protocol (SDP)
   [RFC2327], which is commonly used to describe RTP sessions.  When SDP
   is used to specify sessions employing ES 201 018 DSR codec, the
   mapping is as follows:

   o  The MIME type ("audio") goes in SDP "m=" as the media name.

   o  The MIME subtype ("dsr-es201108") goes in SDP "a=rtpmap" as the
      encoding name.

   o  The optional parameter "rate" also goes in "a=rtpmap" as clock
      rate.

   o  The optional parameters "ptime" and "maxptime" go in the SDP
      "a=ptime" and "a=maxptime" attributes, respectively.








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   Example of usage of ES 201 108 DSR:

      m=audio 49120 RTP/AVP 101
      a=rtpmap:101 dsr-es201108/8000
      a=maxptime:40

6.  Security Considerations

   Implementations using the payload defined in this specification are
   subject to the security considerations discussed in the RTP
   specification [RFC3550] and the RTP profile [RFC3551].  This payload
   does not specify any different security services.

7.  Contributors

   The following individuals contributed to the design of this payload
   format and the writing of this document: Q. Xie (Motorola), D. Pearce
   (Motorola), S. Balasuriya (Motorola), Y. Kim (VerbalTek), S. H. Maes
   (IBM), and, Hari Garudadri (Qualcomm).

8.  Acknowledgments

   The design presented here benefits greatly from an earlier work on
   DSR RTP payload design by Jeff Meunier and Priscilla Walther.  The
   authors also wish to thank Brian Eberman, John Lazzaro, Magnus
   Westerlund, Rainu Pierce, Priscilla Walther, and others for their
   review and valuable comments on this document.

9.  References

9.1  Normative References

   [ES201108]   European Telecommunications Standards Institute (ETSI)
                Standard ES 201 108, "Speech Processing, Transmission
                and Quality Aspects (STQ); Distributed Speech
                Recognition; Front-end Feature Extraction Algorithm;
                Compression Algorithms," Ver. 1.1.2, April 11, 2000.

   [RFC3550]    Schulzrinne, H., Casner, S., Jacobson, V. and R.
                Frederick, "RTP: A Transport Protocol for Real-Time
                Applications", RFC 3550, July 2003.

   [RFC2026]    Bradner, S., "The Internet Standards Process -- Revision
                3", BCP 9, RFC 2026, October 1996.

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




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   [RFC2327]    Handley, M. and V. Jacobson, "SDP: Session Description
                Protocol", RFC 2327, April 1998.

9.2  Informative References

   [RFC3551]    Schulzrinne, H. and S. Casner, "RTP Profile for Audio
                and Video Conferences with Minimal Control", RFC 3551,
                July 2003.

   [RFC2508]    Casner, S. and V. Jacobson, "Compressing IP/UDP/RTP
                Headers for Low-Speed Serial Links", RFC 2508, February
                1999.

   [RFC3095]    Bormann, C., Burmeister, C., Degermark, M., Fukushima,
                H., Hannu, H., Jonsson, L-E, Hakenberg, R., Koren, T.,
                Le, K., Liu, Z., Martensson, A., Miyazaki, A., Svanbro,
                K., Wiebke, T., Yoshimura, T. and H. Zheng, "RObust
                Header Compression (ROHC): Framework and four profiles",
                RFC 3095, July 2001.

10.  IPR Notices

   The IETF takes no position regarding the validity or scope of any
   intellectual property or other rights that might be claimed to
   pertain to the implementation or use of the technology described in
   this document or the extent to which any license under such rights
   might or might not be available; neither does it represent that it
   has made any effort to identify any such rights.  Information on the
   IETF's procedures with respect to rights in standards-track and
   standards-related documentation can be found in BCP-11.  Copies of
   claims of rights made be made available, or the result of an attempt
   made to obtain a general license or permission for the use of such
   proprietary rights by implementors or users of this specification can
   be obtained from the IETF Secretariat.

   The IETF invites any interested party to bring to its attention any
   copyrights, patents or patent applications, or other proprietary
   rights which may cover technology that may be required to practice
   this standard.  Please address the information to the IETF Executive
   Director.











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11.  Authors' Addresses

   David Pearce
   Motorola Labs
   UK Research Laboratory
   Jays Close
   Viables Industrial Estate
   Basingstoke, HANTS, RG22 4PD

   Phone: +44 (0)1256 484 436
   EMail: bdp003@motorola.com


   Senaka Balasuriya
   Motorola, Inc.
   600 U.S Highway 45
   Libertyville, IL 60048, USA

   Phone: +1-847-523-0440
   EMail: Senaka.Balasuriya@motorola.com


   Yoon Kim
   VerbalTek, Inc.
   2921 Copper Rd.
   Santa Clara, CA 95051

   Phone: +1-408-768-4974
   EMail: yoonie@verbaltek.com


   Stephane H. Maes, PhD,
   Oracle
   500 Oracle Parkway, M/S 4op634
   Redwood City, CA 94065 USA

   Phone: +1-650-607-6296.
   EMail: stephane.maes@oracle.com


   Hari Garudadri
   Qualcomm Inc.
   5775, Morehouse Dr.
   San Diego, CA 92121-1714, USA

   Phone: +1-858-651-6383
   EMail: hgarudad@qualcomm.com




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12.  Editor's Address

   Qiaobing Xie
   Motorola, Inc.
   1501 W. Shure Drive, 2-F9
   Arlington Heights, IL 60004, USA

   Phone: +1-847-632-3028
   EMail: Qiaobing.Xie@motorola.com










































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13.  Full Copyright Statement

   Copyright (C) The Internet Society (2003).  All Rights Reserved.

   This document and translations of it may be copied and furnished to
   others, and derivative works that comment on or otherwise explain it
   or assist in its implementation may be prepared, copied, published
   and distributed, in whole or in part, without restriction of any
   kind, provided that the above copyright notice and this paragraph are
   included on all such copies and derivative works.  However, this
   document itself may not be modified in any way, such as by removing
   the copyright notice or references to the Internet Society or other
   Internet organizations, except as needed for the purpose of
   developing Internet standards in which case the procedures for
   copyrights defined in the Internet Standards process must be
   followed, or as required to translate it into languages other than
   English.

   The limited permissions granted above are perpetual and will not be
   revoked by the Internet Society or its successors or assigns.

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