





Network Working Group                                           P. Luthi
Request for Comments: 5577                                      Tandberg
Obsoletes: 3047                                                  R. Even
Category: Standards Track                               Gesher Erove Ltd
                                                               July 2009


          RTP Payload Format for ITU-T Recommendation G.722.1

Abstract

   International Telecommunication Union (ITU-T) Recommendation G.722.1
   is a wide-band audio codec.  This document describes the payload
   format for including G.722.1-generated bit streams within an RTP
   packet.  The document also describes the syntax and semantics of the
   Session Description Protocol (SDP) parameters needed to support
   G.722.1 audio codec.

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) 2009 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 in effect on the date of
   publication of this document (http://trustee.ietf.org/license-info).
   Please review these documents carefully, as they describe your rights
   and restrictions with respect to this document.

   This document may contain material from IETF Documents or IETF
   Contributions published or made publicly available before November
   10, 2008.  The person(s) controlling the copyright in some of this
   material may not have granted the IETF Trust the right to allow
   modifications of such material outside the IETF Standards Process.
   Without obtaining an adequate license from the person(s) controlling
   the copyright in such materials, this document may not be modified
   outside the IETF Standards Process, and derivative works of it may
   not be created outside the IETF Standards Process, except to format
   it for publication as an RFC or to translate it into languages other
   than English.



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Table of Contents

   1. Introduction ....................................................2
   2. Terminology .....................................................3
   3. RTP Usage for G.722.1 ...........................................3
      3.1. RTP G.722.1 Header Fields ..................................3
      3.2. RTP Payload Format for G.722.1 .............................3
      3.3. Multiple G.722.1 Frames in an RTP Packet ...................5
      3.4. Computing the Number of G.722.1 Frames .....................6
   4. IANA Considerations .............................................6
      4.1. Media Type Registration ....................................6
           4.1.1. Registration of Media Type audio/G7221 ..............6
   5. SDP Parameters ..................................................8
      5.1. Usage with the SDP Offer/Answer Model ......................8
   6. Security Considerations .........................................8
   7. Changes from RFC 3047 ...........................................9
   8. Acknowledgments .................................................9
   9. References ......................................................9
      9.1. Normative References .......................................9
      9.2. Informative References ....................................10

1.  Introduction

   ITU-T G.722.1 [ITU.G7221] is a low-complexity coder; it compresses 50
   Hz - 14 kHz audio signals into one of the following bit rates: 24
   kbit/s, 32 kbit/s, or 48 kbit/s.

   The coder may be used for speech, music, and other types of audio.

   Some of the applications for which this coder is suitable are:

   o  Real-time communications such as videoconferencing and telephony

   o  Streaming audio

   o  Archival and messaging

   ITU-T G.722.1 [ITU.G7221] uses 20-ms frames and a sampling rate clock
   of 16 kHz or 32kHz.  The encoding and decoding algorithm can change
   the bit rate at any 20-ms frame boundary, but no bit rate change
   notification is provided in-band with the bit stream.

   For any given bit rate, the number of bits in a frame is a constant.
   Within this fixed frame, G.722.1 uses variable-length coding (e.g.,
   Huffman coding) to represent most of the encoded parameters.  All
   variable-length codes are transmitted in order from the leftmost bit
   (most significant bit -- MSB) to the rightmost bit (least significant
   bit -- LSB), see [ITU.G7221] for more details.



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   The ITU-T standardized bit rates for G.722.1 are 24 kbit/s or
   32kbit/s at 16 Khz sample rate, and 24 kbit/s, 32 kbit/s, or 48
   kbit/s at 32 Khz sample rate.  However, the coding algorithm itself
   has the capability to run at any user-specified bit rate (not just
   24, 32, and 48 kbit/s) while maintaining an audio bandwidth of 50 Hz
   to 14 kHz.  This rate change is accomplished by a linear scaling of
   the codec operation, resulting in frames with size in bits equal to
   1/50 of the corresponding bit rate.

2.  Terminology

   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 RFC 2119 [RFC2119] and
   indicate requirement levels for compliant RTP implementations.

3.  RTP Usage for G.722.1

3.1.  RTP G.722.1 Header Fields

   The RTP header is defined in the RTP specification [RFC3550].  This
   section defines how fields in the RTP header are used.

      Payload Type (PT): The assignment of an RTP payload type for this
      packet format is outside the scope of this document; it is
      specified by the RTP profile under which this payload format is
      used, or it is signaled dynamically out-of-band (e.g., using SDP).

      Marker (M) bit: The M bit is set to zero.

      Extension (X) bit: Defined by the RTP profile used.

      Timestamp: A 32-bit word that corresponds to the sampling instant
      for the first frame in the RTP packet.  The sampling frequency can
      be 16 Khz or 32 Khz.  The RTP timestamp clock frequency of 32 Khz
      SHOULD be used unless only an RTP stream sampled at 16 Khz is
      going to be sent.

3.2.  RTP Payload Format for G.722.1

   The RTP payload for G.722.1 has the format shown in Figure 1.  No
   additional header fields specific to this payload format are
   required.








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      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                               |
      +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
      |                                                               |
      +                 one or more frames of G.722.1                 |
      |                             ....                              |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                     Figure 1: RTP payload for G.722.1.

   Because bit rate is not signaled in-band, a separate out-of-band
   method is REQUIRED to indicate the bit rate (see Section 5 for an
   example of signaling bit rate information using SDP).  For the
   payload format specified here, the bit rate MUST remain constant for
   a particular payload type value.  An application MAY switch bit rates
   and clock rates from packet to packet by defining different payload
   type values and switching between them.

   The use of Huffman coding means that it is not possible to identify
   the various encoded parameters/fields contained within the bit stream
   without first completely decoding the entire frame.  For the purposes
   of packetizing the bit stream in RTP, it is only necessary to
   consider the sequence of bits as output by the G.722.1 encoder and to
   present the same sequence to the decoder.  The payload format
   described here maintains this sequence.

   When operating at 24 kbit/s, 480 bits (60 octets) are produced per
   frame.  When operating at 32 kbit/s, 640 bits (80 octets) are
   produced per frame.  When operating at 48 kbit/s, 960 bits (120
   octets) are produced per frame.  Thus, all three bit rates allow for
   octet alignment without the need for padding bits.

   Figure 2 illustrates how the G.722.1 bit stream MUST be mapped into
   an octet-aligned RTP payload.















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         first bit                                          last bit
         transmitted                                     transmitted
         +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
         |                                                         |
         + sequence of bits (480, 640, or 960) generated by the    |
         |            G.722.1 encoder for transmission             |
         +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


         |           |           |                     |           |
         |           |           |     ...             |           |
         |           |           |                     |           |


         +-+-+-+-+-+-+-+-+-+-+-+-+-+-+ ... +-+-+-+-+-+-+-+-+-+-+-+-+
         |MSB...  LSB|MSB...  LSB|                     |MSB...  LSB|
         +-+-+-+-+-+-+-+-+-+-+-+-+-+-+ ... +-+-+-+-+-+-+-+-+-+-+-+-+
           RTP         RTP                               RTP
           octet 1     octet 2                           octet
                                                      60, 80, 120

           Figure 2:  The G.722.1 encoder bit stream is split into
                      a sequence of octets (60, 80, or 120 depending on
                      the bit rate), and each octet is in turn
                      mapped into an RTP octet.

   When operating at non-standard rates, the payload format MUST follow
   the guidelines illustrated in Figure 2.  It is RECOMMENDED that
   values in the range 16000 to 48000 be used.  Non-standard rates MUST
   have a value that is a multiple of 400 (this maintains octet
   alignment and does not then require (undefined) padding bits for each
   frame if not octet aligned).  For example, a bit rate of 16.4 kbit/s
   will result in a frame of size 328 bits or 41 octets, which is mapped
   into RTP per Figure 2.

3.3.  Multiple G.722.1 Frames in an RTP Packet

   A sender may include more than one consecutive G.722.1 frame in a
   single RTP packet.

   Senders have the following additional restrictions:

   o  Sender SHOULD NOT include more G.722.1 frames in a single RTP
      packet than will fit in the MTU of the RTP transport protocol.

   o  All frames contained in a single RTP packet MUST be of the same
      length and sampled at the same clock rate.  They MUST have the
      same bit rate (octets per frame).



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   o  Frames MUST NOT be split between RTP packets.

   It is RECOMMENDED that the number of frames contained within an RTP
   packet be consistent with the application.  For example, in a
   telephony application where delay is important, the fewer frames per
   packet, the lower the delay; whereas for a delay-insensitive
   streaming or messaging application, many frames per packet would be
   acceptable.

3.4.  Computing the Number of G.722.1 Frames

   Information describing the number of frames contained in an RTP
   packet is not transmitted as part of the RTP payload.  The only way
   to determine the number of G.722.1 frames is to count the total
   number of octets within the RTP packet and divide the octet count by
   the number of expected octets per frame.  This expected octet-per-
   frame count is derived from the bit rate and is therefore a function
   of the payload type.

4.  IANA Considerations

   This document updates the G7221 media type described in RFC 3047.

4.1.  Media Type Registration

   This section describes the media types and names associated with this
   payload format.  The section registers the media types, as per RFC
   4288 [RFC4288]

4.1.1.  Registration of Media Type audio/G7221

   Media type name: audio

   Media subtype name: G7221

   Required parameters:

      bitrate: the data rate for the audio bit stream.  This parameter
      is mandatory because the bit rate is not signaled within the
      G.722.1 bit stream.  At the standard G.722.1 bit rates, the value
      MUST be either 24000 or 32000 at 16 Khz sample rate, and 24000,
      32000, or 48000 at 32 Khz sample rate.  If using the non-standard
      bit rates, then it is RECOMMENDED that values in the range 16000
      to 48000 be used.  Non-standard rates MUST have a value that is a
      multiple of 400 (this maintains octet alignment and does not then
      require (undefined) padding bits for each frame if not octet
      aligned).




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   Optional parameters:

      rate: RTP timestamp clock rate, which is equal to the sampling
      rate.  If the parameter is not specified, a clock rate of 16 Khz
      is assumed.

      ptime: see RFC 4566.  SHOULD be a multiple of 20 ms.

      maxptime: see RFC 4566.  SHOULD be a multiple of 20 ms.

   Encoding considerations:

      This media type is framed and binary, see Section 4.8 in
      [RFC4288].

   Security considerations: See Section 6

   Interoperability considerations:

      Terminals SHOULD offer a media type at 16 Khz sample rate in order
      to interoperate with terminals that do not support the new 32 Khz
      sample rate.

   Published specification: RFC 5577.

   Applications that use this media type:

      Audio and Video streaming and conferencing applications.

   Additional information: none

   Person and email address to contact for further information :

      Roni Even: ron.even.tlv@gmail.com

   Intended usage: COMMON

   Restrictions on usage:

      This media type depends on RTP framing, and hence is only defined
      for transfer via RTP [RFC3550].  Transport within other framing
      protocols is not defined at this time.

   Author: Roni Even

   Change controller:

      IETF Audio/Video Transport working group delegated from the IESG.



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5.  SDP Parameters

   The media types audio/G7221 are mapped to fields in the Session
   Description Protocol (SDP) [RFC4566] as follows:

   o  The media name in the "m=" line of SDP MUST be audio.

   o  The encoding name in the "a=rtpmap" line of SDP MUST be G7221 (the
      media subtype).

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

   o  Only one bitrate SHALL be defined (using the "bitrate=" parameter
      in the a=fmtp line) for each payload type.

5.1.  Usage with the SDP Offer/Answer Model

   When offering G.722.1 over RTP using SDP in an Offer/Answer model
   [RFC3264], the following considerations are necessary.

   The combination of the clock rate in the rtpmap and the bitrate
   parameter defines the configuration of each payload type.  Each
   configuration intended to be used MUST be declared.

   There are two sampling clock rates defined for G.722.1 in this
   document.  RFC 3047 [RFC3047] supports only the 16 Khz clock rate.
   Therefore, a system that wants to use G.722.1 SHOULD offer a payload
   type with clock rate of 16000 for backward interoperability.

   An example of an offer that includes a 16000 and 32000 clock rate is:

             m=audio 49000 RTP/AVP 121 122
             a=rtpmap:121 G7221/16000
             a=fmtp:121 bitrate=24000
             a=rtpmap:122 G7221/32000
             a=fmtp:122 bitrate=48000

6.  Security Considerations

   RTP packets using the payload format defined in this specification
   are subject to the security considerations discussed in the RTP
   specification [RFC3550] and any appropriate RTP profile.  The main
   security considerations for the RTP packet carrying the RTP payload
   format defined within this memo are confidentiality, integrity, and
   source authenticity.  Confidentiality is achieved by encryption of
   the RTP payload.  Integrity of the RTP packets is achieved through a
   suitable cryptographic integrity-protection mechanism.  Such a
   cryptographic system may also allow the authentication of the source



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   of the payload.  A suitable security mechanism for this RTP payload
   format should provide confidentiality, integrity protection, and at
   least source authentication capable of determining if an RTP packet
   is from a member of the RTP session.

   Note that the appropriate mechanism to provide security to RTP and
   payloads following this memo may vary.  It is dependent on the
   application, the transport, and the signaling protocol employed.
   Therefore, a single mechanism is not sufficient; although, if
   suitable, usage of the Secure Real-time Transport Protocol (SRTP) is
   [RFC3711] recommended.  Another mechanism that may be used is IPsec
   [RFC4301] Transport Layer Security (TLS) [RFC5246] (RTP over TCP);
   other alternatives may exist.

   This RTP payload format and its media decoder do not exhibit any
   significant non-uniformity in the receiver-side computational
   complexity for packet processing, and thus are unlikely to pose a
   denial-of-service threat due to the receipt of pathological data.
   Nor does the RTP payload format contain any active content.

7.  Changes from RFC 3047

   This specification obsoletes RFC 3047, adding the support for the
   Superwideband (14 Khz) audio support defined in annex C of the new
   revision of ITU-T G.722.1.

   Other changes:

   Updated the text to be in line with the current rules for RFCs and
   with media type registration conforming to RFC 4288.

8.  Acknowledgments

   The authors would like to thank Tom Taylor for his contribution to
   this work.

9.  References

9.1.  Normative References

   [ITU.G7221]  International Telecommunications Union, "Low-complexity
                coding at 24 and 32 kbit/s for hands-free operation in
                systems with low frame loss", ITU-T Recommendation
                G.722.1, 2005.

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




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   [RFC3264]    Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
                with Session Description Protocol (SDP)", RFC 3264,
                June 2002.

   [RFC3550]    Schulzrinne, H., Casner, S., Frederick, R., and V.
                Jacobson, "RTP: A Transport Protocol for Real-Time
                Applications", STD 64, RFC 3550, July 2003.

   [RFC4566]    Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
                Description Protocol", RFC 4566, July 2006.

9.2.  Informative References

   [RFC3047]    Luthi, P., "RTP Payload Format for ITU-T Recommendation
                G.722.1", RFC 3047, January 2001.

   [RFC3711]    Baugher, M., McGrew, D., Naslund, M., Carrara, E., and
                K. Norrman, "The Secure Real-time Transport Protocol
                (SRTP)", RFC 3711, March 2004.

   [RFC4288]    Freed, N. and J. Klensin, "Media Type Specifications and
                Registration Procedures", BCP 13, RFC 4288,
                December 2005.

   [RFC4301]    Kent, S. and K. Seo, "Security Architecture for the
                Internet Protocol", RFC 4301, December 2005.

   [RFC5246]    Dierks, T. and E. Rescorla, "The Transport Layer
                Security (TLS) Protocol Version 1.2", RFC 5246,
                August 2008.





















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Authors' Addresses

   Patrick Luthi
   Tandberg
   Philip Pedersens vei 22
   1366 Lysaker
   Norway

   EMail: patrick.luthi@tandberg.no


   Roni Even
   Gesher Erove Ltd
   14 David Hamelech
   Tel Aviv  64953
   Israel

   EMail: ron.even.tlv@gmail.com

































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