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Network Working Group                                           P. LuthiRequest for Comments: 3047                                    PictureTelCategory: Standards Track                                   January 2001RTP Payload Format for ITU-T Recommendation G.722.1Status 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 (2001).  All Rights Reserved.Abstract   International Telecommunication Union (ITU-T) Recommendation G.722.1   is a wide-band audio codec, which operates at one of two selectable   bit rates, 24kbit/s or 32kbit/s.  This document describes the payload   format for including G.722.1 generated bit streams within an RTP   packet.  Also included here are the necessary details for the use of   G.722.1 with MIME and SDP.1. Conventions used in this document   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 inRFC-2119 [6].2. Overview of ITU-T Recommendation G.722.1   G.722.1 is a low complexity coder, it compresses 50Hz - 7kHz audio   signals into one of two bit rates, 24 kbit/s or 32 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 messagingLuthi                       Standards Track                     [Page 1]

RFC 3047                 Payload Format G.722.1             January 2001   A fixed frame size of 20 ms is used, and for any given bit rate the   number of bits in a frame is a constant.3. RTP payload format for G.722.1   G.722.1 uses 20 ms frames and a sampling rate clock of 16 kHz, so the   RTP timestamp MUST be in units of 1/16000 of a second.  The RTP   payload for G.722.1 has the format shown in Figure 1.  No additional   header specific to this payload format is required.       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 [3]                           |      +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+      |                                                               |      +                 one or more frames of G.722.1                 |      |                             ....                              |      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+                     Figure 1: RTP payload for G.722.1   The encoding and decoding algorithm can change the bit rate at any   20ms frame boundary, but no bit rate change notification is provided   in-band with the bit stream.  Therefore, a separate out-of-band   method is REQUIRED to indicate the bit rate (seesection 6 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   from packet to packet by defining two payload type values and   switching between them.   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 for a particular class of   applications will assign a payload type for this encoding, or if that   is not done then a payload type in the dynamic range shall be chosen.   The number of bits within a frame is fixed, and within this fixed   frame G.722.1 uses variable length coding (e.g., Huffman coding) to   represent most of the encoded parameters [2].  All variable length   codes are transmitted in order from the left most (most significant -   MSB) bit to the right most (least significant - LSB) bit, see [2] for   more details.   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.Luthi                       Standards Track                     [Page 2]

RFC 3047                 Payload Format G.722.1             January 2001   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 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, and when operating at 32 kbit/s, 640 bits (80 octets) are   produced per frame.  Thus, both 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.   An RTP packet SHALL only contain G.722.1 frames of the same bit rate.      first bit                                          last bit      transmitted                                     transmitted      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+      |                                                         |      + sequence of bits (480 or 640) generated by the          |      |            G.722.1 encoder for transmission             |      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+      |           |           |                     |           |      |           |           |     ...             |           |      |           |           |                     |           |      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+ ... +-+-+-+-+-+-+-+-+-+-+-+-+      |MSB...  LSB|MSB...  LSB|                     |MSB...  LSB|      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+ ... +-+-+-+-+-+-+-+-+-+-+-+-+        RTP         RTP                               RTP        octet 1     octet 2                           octet                                                      60 or 80        Figure 2:  The G.722.1 encoder bit stream is split into                   a sequence of octets (60 or 80 depending on                   the bit rate), and each octet is in turn                   mapped into an RTP octet.   The ITU-T standardized bit rates for G.722.1 are 24 kbit/s and   32kbit/s.  However, the coding algorithm itself has the capability to   run at any user specified bit rate (not just 24 and 32kbit/s) while   maintaining an audio bandwidth of 50 Hz to 7 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.Luthi                       Standards Track                     [Page 3]

RFC 3047                 Payload Format G.722.1             January 2001   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 32000 be used, and that any value MUST   be 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 are mapped into RTP per   Figure 2.3.1 Multiple G.722.1 frames in a RTP packet   More than one G.722.1 frame may be included in a single RTP packet by   a sender.   Senders have the following additional restrictions:   o  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, that is they MUST have the same bit rate (octets per      frame).   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, then 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.3.2 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 (either 60 or 80 per frame,   for 24 kbit/s and 32 kbit/s respectively).4. MIME registration of G.722.1   MIME media type name: audio   MIME subtype: G7221Luthi                       Standards Track                     [Page 4]

RFC 3047                 Payload Format G.722.1             January 2001   Required parameters:         bitrate: the data rate for the audio bit stream.  This         parameter is necessary 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.  If using the         non-standard bit rates, then it is RECOMMENDED that values in         the range 16000 to 32000 be used, and that any value MUST be a         multiple of 400 (this maintains octet alignment and does not         then require (undefined) padding bits for each frame if not         octet aligned).   Optional parameters:         ptime: RECOMMENDED duration of each packet in milliseconds.   Encoding considerations:         This type is only defined for transfer via RTP as specified in         a Work in Progress.   Security Considerations:         SeeSection 6 of RFC 3047.   Interoperability considerations: none   Published specification:         See ITU-T Recommendation G.722.1 [2] for encoding algorithm         details.   Applications which use this media type:         Audio and video streaming and conferencing tools   Additional information: none   Person & email address to contact for further information:         Patrick Luthi         Luthip@pictel.com   Intended usage: COMMON   Author/Change controller:         Author: Patrick Luthi         Change controller: IETF AVT Working GroupLuthi                       Standards Track                     [Page 5]

RFC 3047                 Payload Format G.722.1             January 20015. SDP usage of G.722.1   When conveying information by SDP [5], the encoding name SHALL be   "G7221" (the same as the MIME subtype).  An example of the media   representation in SDP for describing G.722.1 at 24000 bits/sec might   be:         m=audio 49000 RTP/AVP 121         a=rtpmap:121 G7221/16000         a=fmtp:121 bitrate=24000   where "bitrate" is a variable that may take on values of 24000 or   32000 at the standard rates, or values from 16000 to 32000 (and MUST   be an integer multiple of 400) at the non-standard rates.6. Security Considerations   RTP packets using the payload format defined in this specification   are subject to the security considerations discussed in the RTP   specification [3], and any appropriate RTP profile.  This implies   that confidentiality of the media streams is achieved by encryption.   Because the data compression used with this payload format is applied   end-to-end, encryption may be performed after compression so there is   no conflict between the two operations.   A potential denial-of-service threat exists for data encodings using   compression techniques that have non-uniform receiver-end   computational load.  The attacker can inject pathological datagrams   into the stream which are complex to decode and cause the receiver to   be overloaded.  However, this encoding does not exhibit any   significant non-uniformity.   As with any IP-based protocol, in some circumstances a receiver may   be overloaded simply by the receipt of too many packets, either   desired or undesired.  Network-layer authentication may be used to   discard packets from undesired sources, but the processing cost of   the authentication itself may be too high.  In a multicast   environment, pruning of specific sources may be implemented in future   versions of IGMP [7] and in multicast routing protocols to allow a   receiver to select which sources are allowed to reach it.Luthi                       Standards Track                     [Page 6]

RFC 3047                 Payload Format G.722.1             January 20017. References   1. Bradner, S., "The Internet Standards Process -- Revision 3",BCP9,RFC 2026, October 1996.   2. ITU-T Recommendation G.722.1, available online from the ITU      bookstore athttp://www.itu.int.   3. Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson, "RTP:      A Transport Protocol for real-time applications",RFC 1889,      January 1996.  (Updated by a Work in Progress.)   4. Freed, N. and N. Borenstein, "Multipurpose Internet Mail      Extensions (MIME) Part One: Format of Internet Message Bodies",RFC 2045, November 1996.   5. Handley, M. and V. Jacobson, "SDP: Session Description Protocol",RFC 2327, April 1998.   6. Bradner, S., "Key words for use in RFCs to Indicate Requirement      Levels",BCP 14,RFC 2119, March 1997.   7. Deering, S., "Host Extensions for IP Multicasting", STD 5,RFC1112, August 1989.8. Acknowledgments   The author wishes to thank Tony Crossman for starting this work on   G.722.1 packetization and for authoring the initial draft.  The   author also wishes to thank Steve Casner and Colin Perkins for their   valuable feedback and helpful comments.9. Author's Address   Patrick Luthi   PictureTel Corporation   100 Minuteman Road   Andover, MA 01810   USA   Phone: +1 (978) 292 4354   EMail: luthip@pictel.comLuthi                       Standards Track                     [Page 7]

RFC 3047                 Payload Format G.722.1             January 200110. Full Copyright Statement   Copyright (C) The Internet Society (2001).  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.   This document and the information contained herein is provided on an   "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING   TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING   BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION   HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF   MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.Acknowledgement   Funding for the RFC Editor function is currently provided by the   Internet Society.Luthi                       Standards Track                     [Page 8]