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Versions: (draft-ramalho-payload-g7110) 00 01 02 03 04 05 06 RFC 7655

Network Working Group                                    M. Ramalho, Ed.
Internet-Draft                                                  P. Jones
Intended status: Standards Track                           Cisco Systems
Expires: November 12, 2015                                     N. Harada
                                                                     NTT
                                                              M. Perumal
                                                                Ericsson
                                                                 L. Miao
                                                     Huawei Technologies
                                                            May 11, 2015


                     RTP Payload Format for G.711.0
                      draft-ietf-payload-g7110-06

Abstract

   This document specifies the Real-Time Transport Protocol (RTP)
   payload format for ITU-T Recommendation G.711.0.  ITU-T Rec. G.711.0
   defines a lossless and stateless compression for G.711 packet
   payloads typically used in IP networks.  This document also defines a
   storage mode format for G.711.0 and a media type registration for the
   G.711.0 RTP payload format.

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
   Task Force (IETF).  Note that other groups may also distribute
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   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on November 12, 2015.

Copyright Notice

   Copyright (c) 2015 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



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   (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.  Requirements Language . . . . . . . . . . . . . . . . . . . .   3
   3.  G.711.0 Codec Background  . . . . . . . . . . . . . . . . . .   3
     3.1.  General Information and Use of the ITU-T G.711.0 Codec  .   3
     3.2.  Key Properties of G.711.0 Design  . . . . . . . . . . . .   4
     3.3.  G.711 Input Frames to G.711.0 Output Frames . . . . . . .   7
       3.3.1.  Multiple G.711.0 Output Frames per RTP Payload
               Considerations  . . . . . . . . . . . . . . . . . . .   8
   4.  RTP Header and Payload  . . . . . . . . . . . . . . . . . . .   9
     4.1.  G.711.0 RTP Header  . . . . . . . . . . . . . . . . . . .   9
     4.2.  G.711.0 RTP Payload . . . . . . . . . . . . . . . . . . .  10
       4.2.1.  Single G.711.0 Frame per RTP Payload Example  . . . .  11
       4.2.2.  G.711.0 RTP Payload Definition  . . . . . . . . . . .  12
         4.2.2.1.  G.711.0 RTP Payload Encoding Process  . . . . . .  13
       4.2.3.  G.711.0 RTP Payload Decoding Process  . . . . . . . .  14
       4.2.4.  G.711.0 RTP Payload for Multiple Channels . . . . . .  16
   5.  Payload Format Parameters . . . . . . . . . . . . . . . . . .  18
     5.1.  Media Type Registration . . . . . . . . . . . . . . . . .  18
     5.2.  Mapping to SDP Parameters . . . . . . . . . . . . . . . .  21
     5.3.  Offer/Answer Considerations . . . . . . . . . . . . . . .  21
     5.4.  SDP Examples  . . . . . . . . . . . . . . . . . . . . . .  22
       5.4.1.  SDP Example 1 . . . . . . . . . . . . . . . . . . . .  22
       5.4.2.  SDP Example 2 . . . . . . . . . . . . . . . . . . . .  22
   6.  G.711.0 Storage Mode Conventions and Definition . . . . . . .  23
     6.1.  G.711.0 PLC Frame . . . . . . . . . . . . . . . . . . . .  23
     6.2.  G.711.0 Erasure Frame . . . . . . . . . . . . . . . . . .  23
     6.3.  G.711.0 Storage Mode Definition . . . . . . . . . . . . .  24
   7.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  26
   8.  Contributors  . . . . . . . . . . . . . . . . . . . . . . . .  26
   9.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  26
   10. Security Considerations . . . . . . . . . . . . . . . . . . .  26
   11. Congestion Control  . . . . . . . . . . . . . . . . . . . . .  28
   12. References  . . . . . . . . . . . . . . . . . . . . . . . . .  28
     12.1.  Normative References . . . . . . . . . . . . . . . . . .  28
     12.2.  Informative References . . . . . . . . . . . . . . . . .  29
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  30





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1.  Introduction

   The International Telecommunication Union (ITU-T) Recommendation
   G.711.0 [G.711.0] specifies a stateless and lossless compression for
   G.711 packet payloads typically used in Voice over IP (VoIP)
   networks.  This document specifies the Real-Time Transport Protocol
   (RTP) RFC 3550 [RFC3550] payload format and storage modes for this
   compression.

2.  Requirements Language

   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].

3.  G.711.0 Codec Background

   ITU-T Recommendation G.711.0 [G.711.0] is a lossless and stateless
   compression mechanism for ITU-T Recommendation G.711 [G.711] and thus
   is not a "codec" in the sense of "lossy" codecs typically carried by
   RTP.  When negotiated end-to-end ITU-T Rec. G.711.0 is negotiated as
   if it were a codec, with the understanding that ITU-T Rec. G.711.0
   losslessly encoded the underlying (lossy) G.711 pulse code modulation
   (PCM) sample representation of an audio signal.  For this reason
   ITU-T Rec. G.711.0 will be interchangeably referred to in this
   document as a "lossless data compression algorithm" or a "codec",
   depending on context.  Within this document, individual G.711 PCM
   samples will be referred to as "G.711 symbols" or just "symbols" for
   brevity.

   This section describes the ITU-T Recommendation G.711 [G.711] codec,
   its properties, typical uses cases and its key design properties.

3.1.  General Information and Use of the ITU-T G.711.0 Codec

   ITU-T Recommendation G.711 is the benchmark standard for narrowband
   telephony.  It has been successful for many decades because of its
   proven voice quality, ubiquity and utility.  A new ITU-T
   recommendation, G.711.0, has been established for defining a
   stateless and lossless compression for G.711 packet payloads
   typically used in VoIP networks.  ITU-T Rec. G.711.0 is also known as
   ITU-T Rec. G.711 Annex A [G.711-A1], as ITU-T Rec. G.711 Annex A is
   effectively a pointer ITU-T Rec. G.711.0.  Henceforth in this
   document, ITU-T Rec. G.711.0 will simply be referred to as "G.711.0"
   and ITU-T Rec. G.711 simply as "G.711".

   G.711.0 may be employed end-to-end; in which case the RTP payload
   format specification and use is nearly identical to the G.711 RTP



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   specification found in RFC 3551 [RFC3551].  The only significant
   difference for G.711.0 is the required use of a dynamic payload type
   (the static PT of 0 or 8 is presently almost always used with G.711
   even though dynamic assignment of other payload types is allowed) and
   the recommendation not to use Voice Activity Detection (see
   Section 4.1).

   G.711.0, being both lossless and stateless, may also be employed as a
   lossless compression mechanism for G.711 payloads anywhere between
   end systems which have negotiated use of G.711.  Because the only
   significance between the G.711 RTP payload format header and the
   G.711.0 payload format header defined in this document is the payload
   type, a G.711 RTP packet can be losslessly converted to a G.711.0 RTP
   packet simply by compressing the G.711 payload (thus creating a
   G.711.0 payload), changing the payload type to the dynamic value
   desired and copying all the remaining G.711 RTP header fields into
   the corresponding G.711.0 RTP header.  In a similar manner, the
   corresponding decompression of the G.711.0 RTP packet thus created
   back to the original source G.711 RTP packet can be accomplished by
   losslessly decompressing the G.711.0 payload back to the original
   source G.711 payload, changing the payload type back to the payload
   type of the original G.711 RTP packet and copying all the remaining
   G.711.0 RTP header fields into the corresponding G.711 RTP header.
   As a packet produced by the compression and decompression as
   described above is indistinguishable in every detail to the source
   G.711 packet, such compression can be made invisible to the end
   systems.  Specification of how systems on the path between the end
   systems discover each other and negotiate the use of G.711.0
   compression as described in this paragraph is outside the scope of
   this document.

   It is special to note that G.711.0, being both lossless and
   stateless, can be employed multiple times (e.g., on multiple,
   individual hops or series of hops) of a given flow with no
   degradation of quality relative to end-to-end G.711.  Stated another
   way, multiple "lossless transcodes" from/to G.711.0/G.711 do not
   affect voice quality as typically occurs with lossy transcodes to/
   from dissimilar codecs.

   Lastly, it is expected that G.711.0 will be used as an archival
   format for recorded G.711 streams.  Therefore, a G.711.0 Storage Mode
   Format is also included in this document.

3.2.  Key Properties of G.711.0 Design

   The fundamental design of G.711.0 resulted from the desire to
   losslessly encode and compress frames of G.711 symbols independent of
   what types of signals those G.711 frames contained.  The primary



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   G.711.0 use case is for G.711 encoded, zero-mean, acoustic signals
   (such as speech and music).

   G.711.0 attributes are below:

   A1  Compression for zero-mean acoustic signals: G.711.0 was designed
         as its primary use case for the compression of G.711 payloads
         that contained "speech" or other zero-mean acoustic signals.
         G.711.0 obtains greater than 50% average compression in service
         provider environments [ICASSP].

   A2  Lossless for any G.711 payload: G.711.0 was designed to be
         lossless for any valid G.711 payload - even if the payload
         consisted of apparently random G.711 symbols (e.g., a modem or
         FAX payload).  G.711.0 could be used for "aggregate 64 kbps
         G.711 channels" carried over IP without explicit concern if a
         subset of these channels happened to be carrying something
         other than voice or general audio.  To the extent that a
         particular channel carried something other than voice or
         general audio, G.711.0 ensured that it was carried losslessly,
         if not significantly compressed.

   A3  Stateless: Compression of a frame of G.711 symbols was only to be
         dependent on that frame and not on any prior frame.  Although
         greater compression is usually available by observing a longer
         history of past G.711 symbols, it was decided that the
         compression design would be stateless to completely eliminate
         error propagation common in many lossy codec designs (e.g.,
         ITU-T Rec. G.729 [G.729], ITU-T Rec. G.722 [G.722]).  That is,
         the decoding process need not be concerned about lost prior
         packets because the decompression of a given G.711.0 frame is
         not dependent on potentially lost prior G.711.0 frames.  Owing
         to this stateless property, the frames input to the G.711.0
         encoder may be changed "on-the-fly" (a 5 ms encoding could be
         followed by a 20 ms encoding).

   A4  Self-describing: This property is defined as the ability to
         determine how many source G.711 samples are contained within
         the G.711.0 frame solely by information contained within the
         G.711.0 frame.  Generally, the number of source G.711 symbols
         can be determined by decoding the initial octets of the
         compressed G.711.0 frame (these octets are called "prefix
         codes" in the standard).  A G.711.0 decoder need not know how
         many symbols are contained in the original G.711 frame (e.g.,
         parameter ptime in Session Description Protocol, SDP,
         [RFC4566]), as it is able to decompress the G.711.0 frame
         presented to it without signaling knowledge.




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   A5  Accommodate G.711 payload sizes typically used in IP: G.711 input
         frames of length typically found in VoIP applications represent
         SDP ptime values of 5 ms, 10 ms, 20 ms, 30 ms or 40 ms.  Since
         the dominant sampling frequency for G.711 is 8000 samples per
         second, G.711.0 was designed to compress G.711 input frames of
         40, 80, 160, 240 or 320 samples.

   A6  Bounded expansion: Since attribute A2 above requires G.711.0 to
         be lossless for any payload (which could consist of any
         combination of octets with each octet spanning the entire space
         of 2^8 values), by definition there exists at least one
         potential G.711 payload which must be "uncompressible".  Since
         the quantum of compression is an octet, the minimum expansion
         of such an uncompressible payload was designed to be the
         minimum possible of one octet.  Thus G.711.0 "compressed"
         frames can be of length one octet to X+1 octets, where X is the
         size of the input G.711 frame in octets.  G.711.0 can therefore
         be viewed as a Variable Bit Rate (VBR) encoding in which the
         size of the G.711.0 output frame is a function of the G.711
         symbols input to it.

   A7  Algorithmic delay: G.711.0 was designed to have the algorithmic
         delay equal to the time represented by the number of samples in
         the G.711 input frame (i.e., no "look-ahead").

   A8  Low Complexity: Less than 1.0 Weighted Million Operations Per
         Second (WMOPS) average and low memory footprint (~5k octets
         RAM, ~5.7k octets ROM and ~3.6 basic operations) [ICASSP]
         [G.711.0].

   A9  Both A-law and mu-law supported: G.711 has two operating laws,
         A-law and mu-law.  These two laws are also known as PCMA and
         PCMU in RTP applications RFC 3551 [RFC3551].

   These attributes generally make it trivial to compress a G.711 input
   frame consisting of 40, 80, 160, 240 or 320 samples.  After the input
   frame is presented to a G.711.0 encoder, a G.711.0 "self-describing"
   output frame is produced.  The number of samples contained within
   this frame is easily determined at the G.711.0 decoder by virtue of
   attribute A4.  The G.711.0 decoder can decode the G.711.0 frame back
   to a G.711 frame by using only data within the G.711.0 frame.

   Lastly we note that losing a G.711.0 encoded packet is identical in
   effect of losing a G.711 packet (when using RTP); this is because a
   G.711.0 payload, like the corresponding G.711 payload, is stateless.
   Thus, it is anticipated that existing G.711 PLC mechanisms will be
   employed when a G.711.0 packet is lost and an identical MOS
   degradation relative to G.711 loss will be achieved.



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3.3.  G.711 Input Frames to G.711.0 Output Frames

   G.711.0 is a lossless and stateless compression of G.711 frames.  The
   following figure depicts this where "A" is the process of G.711.0
   encoding and "B" is the process of G.711.0 decoding.

        1:1 Mapping from G.711 Input Frame to G.711.0 Output Frame


    |--------------------------|  A   |------------------------------|
    |    G.711 Input Frame     |----->|     G.711.0 Output Frame     |
    |       of X Octets        |      |  containing 1 to X+1 Octets  |
    | (where X MUST be 40, 80, |      | (precise value dependent on  |
    | 160, 240 or 320 octets)  |<-----| G.711.0 ability to compress) |
    |__________________________|  B   |______________________________|


                                 Figure 1

   Note that the mapping is 1:1 (lossless) in both directions, subject
   to two constraints.  The first constraint is that the input frame
   provided to the G.711.0 encoder (process "A") has a specific number
   of input G.711 symbols consistent with attribute A5 (40, 80, 160, 240
   or 320 octets).  The second constraint is that the companding law
   used to create the G.711 input frame (A-law or mu-law) must be known,
   consistent with attribute A9.

   Subject to these two constraints, the input G.711 frame is processed
   by the G.711.0 encoder ("process A") and produces a "self-describing"
   G.711.0 output frame, consistent with attribute A4.  Depending on the
   source G.711 symbols, the G.711.0 output frame can contain anywhere
   from 1 to X+1 octets, where X is the number of input G.711 symbols.
   Compression results for virtually every zero-mean acoustic signal
   encoded by G.711.0.

   Since the G.711.0 output frame is "self-describing", a G.711.0
   decoder (process "B") can losslessly reproduce the original G.711
   input frame with only the knowledge of which companding law was used
   (A-law or mu-law).  The first octet of a G.711.0 frame is called the
   "Prefix Code" octet; the information within this octet conveys how
   many G.711 symbols the decoder is to create from a given G.711.0
   input frame (i.e., 0, 40, 80, 160, 240 or 320).  The Prefix Code
   value of 0x00 is used to denote zero G.711 source symbols, which
   allows the use of 0x00 as a payload padding octet (to be described
   later in Section 3.3.1).

   Since G.711.0 was designed with typical G.711 payload lengths as a
   design constraint (attribute A5), this lossless encoding can be



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   performed only with knowledge of the companding law being used.  This
   information is anticipated to be signaled in SDP and will be
   described later in this document.

   If the original inputs were known to be from a zero-mean acoustic
   signal coded by G.711, an intelligent G.711.0 encoder could infer the
   G.711 companding law in use (via G.711 input signal amplitude
   histogram statistics).  Likewise, an intelligent G.711.0 decoder
   producing G.711 from the G.711.0 frames could also infer which
   encoding law in use.  Thus G.711.0 could be designed for use in
   applications that have limited stream signaling between the G.711
   endpoints (i.e., they only know "G.711 at 8k sampling is being used",
   but nothing more).  Such usage is not further described in this
   document.  Additionally, if the original inputs were known to come
   from zero-mean acoustic signals, an intelligent G.711.0 encoder could
   tell if the G.711.0 payload had been encrypted - as the symbols would
   not have the distribution expected in either companding law and would
   appear random.  Such determination is also not further discussed in
   this document.

   It is easily seen that this process is 1:1 and that G.711.0 based
   lossless compression can be employed multiple times, as the original
   G.711 input symbols are always reproduced with 100% fidelity.

3.3.1.  Multiple G.711.0 Output Frames per RTP Payload Considerations

   As a general rule, G.711.0 frames containing more source G.711
   symbols (from a given channel) will typically result in higher
   compression, but there are exceptions to this rule.  A G.711.0
   encoder may choose to encode 20 ms of input G.711 symbols as: 1) a
   single 20 ms G.711.0 frame, or 2) as two 10 ms G.711.0 frames, or 3)
   any other combination of 5 ms or 10 ms G.711.0 frames - depending on
   which encoding resulted in fewer bits.  As an example, an intelligent
   encoder might encode 20 ms of G.711 symbols as two 10 ms G.711.0
   frames if the first 10 ms was "silence" and two G.711.0 frames took
   fewer bits than any other possible encoding combination of G.711.0
   frame sizes.

   During the process of G.711.0 standardization it was recognized that
   although it is sometimes advantageous to encode integer multiples of
   40 G.711 symbols in whatever input symbol format resulted in the most
   compression (as per above), the simplest choice is to encode the
   entire ptime's worth of input G.711 symbols into one G.711.0 frame
   (if the ptime supported it).  This is especially so since the larger
   number of source G.711 symbols typically resulted in the highest
   compression anyway and there is added complexity in searching for
   other possibilities (involving more G.711.0 frames) which were
   unlikely to produce a more bit efficient result.



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   The design of ITU-T Rec. G.711.0 [G.711.0] foresaw the possibility of
   multiple G.711.0 input frames in that the decoder was defined to
   decode what it refers to as an incoming "bit stream".  For this
   specification, the bit stream is the G.711.0 RTP payload itself.
   Thus, the decoder will take the G.711.0 RTP payload and will produce
   an output frame containing the original G.711 symbols independent of
   how many G.711.0 frames were present in it.  Additionally, any number
   of 0x00 padding octets placed between the G.711.0 frames will be
   silently (and safely) ignored by the G.711.0 decoding process
   Section 4.2.3).

   To recap, a G.711.0 encoder may choose to encode incoming G.711
   symbols into one or more than one G.711.0 frames and put the
   resultant frame(s) into the G.711.0 RTP payload.  Zero or more 0x00
   padding octets may also be included in the G.711.0 RTP payload.  The
   G.711.0 decoder, being insensitive to the number of G.711.0 encoded
   frames that are contained within it, will decode the G.711.0 RTP
   payload into the source G.711 symbols.  Although examples of single
   or multiple G.711 frame cases will be illustrated in Section 4.2, the
   multiple G.711.0 frame cases MUST be supported and there is no need
   for negotiation (SDP or otherwise) required for it.

4.  RTP Header and Payload

   In this section we describe the precise format for G.711.0 frames
   carried via RTP.  We begin with RTP header description relative to
   G.711, then provide two G.711.0 payload examples.

4.1.  G.711.0 RTP Header

   Relative to G.711 RTP headers, the utilization of G.711.0 does not
   create any special requirements with respect to the contents of the
   RTP packet header.  The only significant difference is that the
   payload type (PT) RTP header field MUST have a value corresponding to
   the dynamic payload type assigned to the flow.  This is in contrast
   to most current uses of G.711 which typically use the static payload
   assignment of PT = 0 (PCMU) or PT = 8 (PCMA) [RFC3551] even though
   the negotiation and use of dynamic payload types is allowed for
   G.711.  With the exception of rare PT exhaustion cases, the existing
   G.711 PT values of 0 and 8 MUST NOT be used for G.711.0 (helping to
   avoid possible payload confusion with G.711 payloads).

   Voice Activity Detection (VAD) SHOULD NOT be used when G.711.0 is
   negotiated because G.711.0 obtains high compression during "VAD
   silence intervals" and one of the advantages of G.711.0 over G.711
   with VAD is the lack of any VAD-inducing artifacts in the received
   signal.  However, if VAD is employed, the Marker bit (M) MUST be set
   in the first packet of a talkspurt (the first packet after a silence



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   period in which packets have not been transmitted contiguously as per
   rules specified in [RFC3551] for G.711 payloads).  This definition,
   being consistent with the G.711 RTP VAD use, further allows lossless
   transcoding between G.711 RTP packets and G.711.0 RTP packets as
   described in Section 3.1.

   With this introduction, the RTP packet header fields are defined as
   follows:

      V - As per [RFC3550]

      P - As per [RFC3550]

      X - As per [RFC3550]

      CC - As per [RFC3550]

      M - As per [RFC3550] and [RFC3551]

      PT - The assignment of an RTP payload type for the format defined
      in this memo is outside the scope of this document.  The RTP
      profiles in use currently mandate binding the payload type
      dynamically for this payload format (see [RFC3550], [RFC4585]).

      SN - As per [RFC3550]

      timestamp - As per [RFC3550]

      SSRC - As per [RFC3550]

      CSRC - As per [RFC3550]

   Where V (version bits), P (padding bit), X (extension bit), CC (CSRC
   count), M (marker bit), PT (payload type), SN (sequence number),
   timestamp, SSRC (synchronizing source) and CSRC (contributing
   sources) are as defined in [RFC3550] and as typically used with
   G.711.  PT (payload type) is as defined in [RFC3551].

4.2.  G.711.0 RTP Payload

   This section defines the G.711.0 RTP payload and illustrates it by
   means of two examples.

   The first example, in Section 4.2.1, depicts the case when it is
   desired to carry only one G.711.0 frame in the RTP payload.  This
   case is expected to be the dominant use case and is shown separately
   for the purposes of clarity.




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   The second example, in Section 4.2.2, depicts the general case when
   it is desired to carry one or more G.711.0 frames in the RTP payload.
   This is the actual definition of the G.711.0 RTP payload.

4.2.1.  Single G.711.0 Frame per RTP Payload Example

   This example depicts a single G.711.0 frame in the RTP payload.  This
   is expected to be the dominant RTP payload case for G.711.0, as the
   G.711.0 encoding process supports the SDP packet times (ptime and
   maxptime, see [RFC4566]) commonly used when G.711 is transported in
   RTP.  Additionally, as mentioned previously, larger G.711.0 frames
   generally compress more effectively than a multiplicity of smaller
   G.711.0 frames.

   The following Figure illustrates the single G.711.0 frame per RTP
   payload case.

                 Single G.711.0 Frame in RTP Payload Case


                 |-------------------|-------------------|
                 | One G.711.0 Frame | Zero or more 0x00 |
                 |                   |   Padding Octets  |
                 |___________________|___________________|


                                 Figure 2

   Encoding Process: A single G.711.0 frame is inserted into the RTP
   payload.  The amount of time represented by the G.711 symbols
   compressed in the G.711.0 frame MUST correspond to the ptime signaled
   for applications using SDP.  Although generally not desired, padding
   desired in the RTP payload after the G.711.0 frame MAY be created by
   placing one or more 0x00 octets after the G.711.0 frame.  Such
   padding may be desired based on security considerations (see
   Section 10).

   Decoding Process: Passing the entire RTP payload to the G.711.0
   decoder is sufficient for the G.711.0 decoder to create the source
   G.711 symbols.  Any padding inserted after the G.711.0 frame (i.e.,
   the 0x00 octets) present in the RTP payload is silently ignored by
   the G.711.0 decoding process.  The decoding process is fully
   described in Section 4.2.3 below.








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4.2.2.  G.711.0 RTP Payload Definition

   This section defines the G.711.0 RTP payload and illustrates the case
   of when one or more G.711.0 frames are to be placed in the payload.
   All G.711.0 RTP decoders MUST support the general case described in
   this section (rationale presented previously in Section 3.3.1).

   Note that since each G.711.0 frame is self-describing (see Attribute
   A4 in Section 3.2), the individual G.711.0 frames in the RTP payload
   need not represent the same duration of time (i.e., a 5 ms G.711.0
   frame could be followed by a 20 ms G.711.0 frame).  Owing to this,
   the amount of time represented in the RTP payload MAY be any integer
   multiple of 5 ms (as 5 ms is the smallest interval of time that can
   be represented in a G.711.0 frame).

   The following Figure illustrates the one or more G.711.0 frames per
   RTP payload case where the number of G.711.0 frames placed in the RTP
   payload is N.  We note that when N is equal to 1 that this case is
   identical to the previous example.

              One or More G.711.0 Frames in RTP Payload Case


       |----------|---------|----------|---------|----------------|
       | First    | Second  |          | Nth     | Zero or more   |
       | G.711.0  | G.711.0 |   ...    | G.711.0 |     0x00       |
       | Frame    | Frame   |          | Frame   | Padding Octets |
       |__________|_________|__________|_________|________________|


                                 Figure 3

   We note here that when we have multiple G.711.0 frames that the
   individual frames can be, and generally are, of different lengths.
   The decoding process described in Section 4.2.3 is used to determine
   the frame boundaries.

   Encoding Process: One or more G.711.0 frames are placed in the RTP
   payload simply by concatenating the G.711.0 frames together.  The
   amount of time represented by the G.711 symbols compressed in all the
   G.711.0 frames in the RTP payload MUST correspond to the ptime
   signaled for applications using SDP.  Although not generally desired,
   padding in the RTP payload SHOULD be placed after the last G.711.0
   frame in the payload and MAY be created by placing one or more 0x00
   octets after the last G.711.0 frame.  Such padding may be desired
   based on security considerations (see Section 10).  Additional
   encoding process details and considerations are specified later in
   Section 4.2.2.1.



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   Decoding Process: As G.711.0 frames can be of varying length, the
   payload decoding process described in Section 4.2.3 is used to
   determine where the individual G.711.0 frame boundaries are.  Any
   padding octets inserted before or after any G.711.0 frame in the RTP
   payload is silently (and safely) ignored by the G.711.0 decoding
   process specified in Section 4.2.3.

4.2.2.1.  G.711.0 RTP Payload Encoding Process

   ITU-T G.711.0 supports five possible input frame lengths: 40, 80,
   160, 240, and 320 samples per frame and the rationale for choosing
   those lengths was given in the description of property A5 in
   Section 3.2.  Assuming 8000 sample per second, these lengths
   correspond to input frames representing 5 ms, 10 ms, 20 ms, 30 ms or
   40 ms.  So while the standard assumed the input "bit stream"
   consisted of G.711 symbols of some integer multiple of 5 ms in
   length, it did not specify exactly what frame lengths to use as input
   to the G.711.0 encoder itself.  The intent of this section is to
   provide some guidance for the selection.

   Consider a typical IETF use case of 20 ms (160 octets) of G.711 input
   samples represented in a G.711.0 payload and signaled by using the
   SDP parameter ptime.  As described in Section 3.3.1, the simplest way
   to encode these 160 octets is to pass the entire 160 octet to the
   G.711.0 encoder, resulting in precisely one G.711.0 compressed frame,
   and put that singular frame into the G.711.0 RTP payload.  However,
   neither the ITU-T G.711.0 standard nor this IETF payload format
   mandates this.  In fact 20 ms of input G.711 symbols can be encoded
   as 1, 2, 3 or 4 G.711.0 frames in any one of six combinations (i.e.,
   {20ms}, {10ms:10ms}, {10ms:5ms:5ms}, {5ms:10ms:5ms}, {5ms:5ms:10ms},
   {5ms:5ms:5ms:5ms}) and any of these combinations would decompress
   into the same source 160 G.711 octets.  As an aside, we note that the
   first octet of any G.711.0 frame will be the prefix code octet and
   information in this octet determines how many G.711 symbols are
   represented in the G.711.0 frame.

   Notwithstanding the above, we expect one of two encodings to be used
   by implementers: the simplest possible (one 160 byte input to the
   G.711.0 encoder which usually results in the highest compression) or
   the combination of possible input frames to a G.711.0 encoder that
   resulted in the highest compression for the payload.  The explicit
   mention of this issue in this IETF document was deemed important
   because the ITU-T G.711.0 standard is silent on this issue and there
   is a desire for this issue to be documented in a formal Standards
   Developing Organization (SDO) document (i.e., here).






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4.2.3.  G.711.0 RTP Payload Decoding Process

   The G.711.0 decoding process is a standard part of G.711.0 bit stream
   decoding and is implemented in the ITU-T Rec. G.711.0 reference code.
   The decoding process algorithm described in this section is a slight
   enhancement of the ITU-T reference code to explicitly accommodate RTP
   padding (as described above).

   Before describing the decoding, we note here that the largest
   possible G.711.0 frame is created whenever the largest number of
   G.711 symbols is encoded (320 from Section 3.2, property A5) and
   these 320 symbols are "uncompressible" by the G.711.0 encoder.  In
   this case (via property A6 in Section 3.2) the G.711.0 output frame
   will be 321 octets long.  We also note that the value 0x00 chosen for
   the optional padding cannot be the first octet of a valid ITU-T Rec.
   G.711.0 frame (see [G.711.0]).  We also note that whenever more than
   one G.711.0 frame is contained in the RTP payload, the decoding of
   the individual G.711.0 frames will occur multiple times.

   For the decoding algorithm below, let N be the number of octets in
   the RTP payload (i.e., excluding any RTP padding, but including any
   RTP payload padding), let P equal the number of RTP payload octets
   processed by the G.711.0 decoding process, let K be the number of
   G.711 symbols presently in the output buffer, let Q be the number of
   octets contained in the G.711.0 frame being processed and let "!="
   represent not equal to.  The keyword "STOP" is used below to indicate
   the end of the processing of G.711.0 frames in the RTP payload.  The
   algorithm below assumes an output buffer for the decoded G.711 source
   symbols of length sufficient to accommodate the expected number of
   G.711 symbols and an input buffer of length 321 octets.

   G.711.0 RTP Payload Decoding Heuristic:

   H1  Initialization of counters: Initialize P, the number of processed
         octets counter, to zero.  Initialize K, the counter for how
         many G.711 symbols are in the output buffer, to zero.
         Initialize N to the number of octets in the RTP payload
         (including any RTP payload padding).  Go to H2.

   H2  Read internal buffer: Read min{320+1, (N-P)-1} octets into the
         internal buffer from the (P+1) octet of the RTP payload.  We
         note at this point, N-P octets have yet to be processed and
         that 320+1 octets is the largest possible G.711.0 frame.  Also
         note that in the common case of zero-based array indexing of a
         uint8 array of octets, that this operation will read octets
         from index P through index [min{320+1, (N-P)}] from the RTP
         payload.  Go to H3.




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   H3  Analyze the first octet in the internal buffer: If this octet
         0x00 (a padding octet) go to H4, otherwise go to H5 (process a
         G.711.0 frame).

   H4  Process padding octet (no G.711 symbols generated): Increment the
         processed packets counter by one (set P = P + 1).  If the
         result of this increment results in P >= N then STOP (as all
         RTP Payload octets have been processed), otherwise go to H2.

   H5  Process an individual G.711.0 frame (produce G.711 samples in the
         output frame): Pass the internal buffer to the G.711.0 decoder.
         The G.711.0 decoder will read the first octet (called the
         "prefix code" octet in ITU-T Rec. G.711.0 [G.711.0]) to
         determine the number of source G.711 samples M are contained in
         this G.711.0 frame.  The G.711.0 decoder will produce exactly M
         G.711 source symbols (M can only have values of 0, 40, 80, 160,
         240 or 320).  If K = 0, these M symbols will be the first in
         the output buffer and are placed at the beginning of the output
         buffer.  If K != 0, concatenate these M symbols with the prior
         symbols in the output buffer (there are K prior symbols in the
         buffer).  Set K = K + M (as there are now this many G.711
         source symbols in the output buffer).  The G.711.0 decoder will
         have consumed some number of octets, Q, in the internal buffer
         to produce the M G.711 symbols.  Increment the number of
         payload octet processed counter by this quantity (set P = P +
         Q).  If the result of this increment results in P >= N then
         STOP (as all RTP Payload octets have been processed), otherwise
         go to H2.

   At this point, the output buffer will contain precisely K G.711
   source symbols which should correspond to the ptime signaled if SDP
   was used and the encoding process was without error.  If ptime was
   signaled via SDP and the number of G.711 symbols in the output buffer
   is other than what corresponds to ptime, the packet MUST be discarded
   unless other system design knowledge allows for otherwise (e.g.,
   occasional 5 ms clock slips causing one more or one less G.711.0
   frame than nominal to be in the payload).  Lastly, due to the buffer
   reads in H2 being bounded (to 321 octets or less), N being bounded to
   the size of the G.711.0 RTP payload, and M being bounded to the
   number of source G.711 symbols, there is no buffer overrun risk.

   We also note, as an aside, that the algorithm above (and the ITU-T
   G.711.0 reference code) accommodates padding octets (0x00) placed
   anywhere between G.711.0 frames in the RTP payload as well as prior
   to or after any or all G.711.0 frames.  The ITU-T G.711.0 reference
   code does not have Step H3 and H4 as separate steps (i.e., Step H5
   immediately follows H2) at the added computational cost of some
   additional buffer passing to/from the G.711.0 frame decoder



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   functions.  That is the G.711.0 decoder in the reference code
   "silently ignores" 0x00 padding octets at the beginning of what it
   believes to be a G.711.0 encoded frame boundary.  Thus Step H3 and
   Step H4 above are an optimization over the reference code shown for
   clarity.

   If the decoder is at a playout endpoint location, this G.711 buffer
   SHOULD be used in the same manner as a received G.711 RTP payload
   would have been used (passed to a playout buffer, to a PLC
   implementation, etc.).

   We explicitly note that a framing error condition will result
   whenever the buffer sent to a G.711.0 decoder does not begin with a
   valid first G.711.0 frame octet (i.e., a valid G.711.0 prefix code or
   a 0x00 padding octet).  The expected result is that the decoder will
   not produce the desired/correct G.711 source symbols.  However, as
   already noted, the output returned by the G.711.0 decoder will be
   bounded (to less than 321 octets per G.711.0 decode request) and if
   the number of the (presumed) G.711 symbols produced is known to be in
   error, the decoded output MUST be discarded.

4.2.4.  G.711.0 RTP Payload for Multiple Channels

   In this section we describe the use of multiple "channels" of G.711
   data encoded by G.711.0 compression.

   The dominant use of G.711 in RTP transport has been for single
   channel use cases.  For this case, the above G.711.0 encoding and
   decoding process is used.  However, the multiple channel case for
   G.711.0 (a frame-based compression) is different from G.711 (a
   sample-based encoding) and is described separately here.

   RFC 3551 [RFC3551] provides guidelines for encoding audio channels
   (Section 4) and for the ordering of the channels within the RTP
   payload (Section 4.1).  The ordering guidelines in RFC 3551,
   Section 4.1 SHOULD be used unless an application-specific channel
   ordering is more appropriate.

   An implicit assumption in RFC 3551 is that all the channel data
   multiplexed into a RTP payload MUST represent the same physical time
   span.  The case for G.711.0 is no different; the underlying G.711
   data for all channels in a G.711.0 RTP payload MUST span the same
   interval in time (e.g., the same "ptime" for a SDP-specified codec
   negotiation).

   RFC 3551 provides guidelines for sample-based encodings such as G.711
   in Section 4.2.  This guidance is tantamount to interleaving the




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   individual samples in that they SHOULD be packed in consecutive
   octets.

   RFC 3551 provides guidelines for frame-based encodings in which the
   frames are interleaved.  However, this guidance stems from the
   assumption that "the frame size for frame-oriented codecs is a
   given".  However, this assumption is not valid for G.711.0 in that
   individual consecutive G.711.0 frames (as per Section 4.2.2) can:

      1) represent different time spans (e.g., two 5 ms G.711.0 frames
      in lieu of one 10 ms G.711.0 frame), and

      2) be of different lengths in octets (and typically are).

   Therefore a different, but also simple, concatenation-based approach
   is specified in this RFC.

   For the multiple channel G.711.0 case, each G.711 channel is
   independently encoded into one or more G.711.0 frames defined here as
   a "G.711.0 channel superframe".  Each one of these superframes is
   identical to the multiple G.711.0 frame case illustrated in Figure 3
   of Section 4.2.2 in which each superframe can have one or more
   individual G.711.0 frames within it.  Then each G.711.0 channel
   superframe is concatenated - in channel order - into a G.711.0 RTP
   payload.  Then, if optional G.711.0 padding octets (0x00) are
   desired, it is RECOMMENDED that these octets are placed after the
   last G.711.0 channel superframe.  As per above, such padding may be
   desired based on security considerations (see Section 10).  This is
   depicted in the following Figure 4 below.

            Multiple G.711.0 Channel Superframes in RTP Payload


           |----------|---------|----------|---------|---------|
           | First    | Second  |          | Nth     | Zero    |
           | G.711.0  | G.711.0 |   ...    | G.711.0 | or more |
           | Channel  | Channel |          | Channel | 0x00    |
           | Super-   | Super-  |          | Super   | Padding |
           | Frame    | Frame   |          | Frame   | Octets  |
           |__________|_________|__________|_________|_________|


                                 Figure 4

   We note that although the individual superframes can be of different
   lengths in octets (and usually are), that the number of G.711 source
   symbols represented - in compressed form - in each channel superframe




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   is identical (since all the channels represent the identically same
   time interval).

   The G.711.0 decoder at the receiving end simply decodes the entire
   G.711.0 (multiple channel) payload into individual G.711 symbols.  If
   M such G.711 symbols result and there were N channels, then the first
   M/N G.711 samples would be from the first channel, the second M/N
   G.711 samples would be from the second channel, and so on until the
   Nth set of G.711 samples are found.  Similarly, if the number of
   channels was not known, but the payload "ptime" was known, one could
   infer (knowing the sampling rate) how many G.711 symbols each channel
   contained; then with this knowledge determine how many channels of
   data were contained in the payload.  When SDP is used, the number of
   channels is known because the optional parameter is a MUST when there
   is more than one channel negotiated (see Section 5.1).  Additionally,
   when SDP is used the parameter ptime is a RECOMMENDED optional
   parameter.  We note that if both parameters channels and ptime are
   known that one could provide a check for the other and the converse.
   Whichever algorithm is used to determine the number of channels, if
   the length of the source G.711 symbols in the payload (M) is not an
   integer multiple of the number of channels (N), then the packet
   SHOULD be discarded.

   Lastly we note that although any padding for the multiple channel
   G.711.0 payload is RECOMMENDED to be placed at the end of the
   payload, the G.711.0 decoding algorithm described in Section 4.2.3
   will successfully decode the payload in Figure 4 if the 0x00 padding
   octet is placed anywhere before or after any individual G.711.0 frame
   in the RTP payload.  The number of padding octets introduced at any
   G.711.0 frame boundary therefore does not affect the number M of the
   source G.711 symbols produced.  Thus the decision for padding MAY be
   made on a per-superframe basis.

5.  Payload Format Parameters

   This section defines the parameters that may be used to configure
   optional features in the G.711.0 RTP transmission.

   The parameters defined here are a part of the media subtype
   registration for the G.711.0 codec.  Mapping of the parameters into
   Session Description Protocol (SDP) RFC 4566 [RFC4566] is also
   provided for those applications that use SDP.

5.1.  Media Type Registration

   Type name: audio

   Subtype name: G711-0



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   Required parameters:

      clock rate: The RTP timestamp clock rate, which is equal to the
      sampling rate.  The typical rate used with G.711 encoding is 8000,
      but other rates may be specified.  The default rate is 8000.

      complaw: This format specific parameter, specified on the "a=fmtp:
      line", indicates the companding law (A-law or mu-law) employed.
      This format specific parameter, as per RFC 4566 [RFC4566], is
      given unchanged to the media tool using this format.  The case-
      insensitive values are "complaw=al" or "complaw=mu" are used for
      A-law and mu-law, respectively.

   Optional parameters:

      channels: See RFC 4566 [RFC4566] for definition.  Specifies how
      many audio streams are represented in the G.711.0 payload and MUST
      be present if the number of channels is greater than one.  This
      parameter defaults to 1 if not present (as per RFC 4566) and is
      typically a non-zero small-valued positive integer.  It is
      expected that implementations that specify multiple channels will
      also define a mechanism to map the channels appropriately within
      their system design, otherwise the channel order specified in RFC
      3551 [RFC3551] Section 4.1 will be assumed (e.g., left, right,
      center, ... ).  Similar to the usual interpretation in RFC 3551
      [RFC3551], the number of channels SHALL be a non-zero positive
      integer.

      maxptime: See RFC 4566 [RFC4566] for definition.

      ptime: See RFC 4566 [RFC4566] for definition.  The inclusion of
      "ptime" is RECOMMENDED and SHOULD be in the SDP unless there is an
      application specific reason not to include it (e.g., an
      application that has a variable ptime on a packet-by-packet
      basis).  For constant ptime applications, it is considered good
      form to include "ptime" in the SDP for session diagnostic
      purposes.  For the constant ptime multiple channel case described
      in Section 4.2.2, the inclusion of "ptime" can provide a desirable
      payload check.

   Encoding considerations:

      This media type is framed binary data (see Section 4.8 in RFC 6838
      [RFC6838]) compressed as per ITU-T Rec. G.711.0.

   Security considerations:

      See Section 10.



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   Interoperability considerations: none

   Published specification:

      ITU-T Rec. G.711.0 and RFC XXXX.

      [ RFC Editor: please replace XXXXX with a reference to this RFC ]

   Applications that use this media type:

      Although initially conceived for VoIP, the use of G.711.0, like
      G.711 before it, may find use within audio and video streaming
      and/or conferencing applications for the audio portion of those
      applications.

   Additional information:

   The following applies to stored-file transfer methods:

         Magic numbers: #!G7110A\n or #!G7110M\n (for A-law or MU-law
         encodings respectively, see Section 6).

         File Extensions: None

         Macintosh file type code: None

         Object identifier or OIL: None

   Person & email address to contact for further information:

      Michael A.  Ramalho <mramalho@cisco.com> or <mar42@cornell.edu>

   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: Michael A.  Ramalho

   Change controller:

      IETF Payload working group delegated from the IESG.






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5.2.  Mapping to SDP Parameters

   The information carried in the media type specification has a
   specific mapping to fields in the Session Description Protocol (SDP),
   which is commonly used to describe a RTP session.  When SDP is used
   to specify sessions employing G.711.0, the mapping is as follows:

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

   o  The media subtype ("G711-0") goes in SDP "a=rtpmap" as the
      encoding name.

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

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

   o  Remaining parameters go in the SDP "a=fmtp" attribute by copying
      them directly from the media type string as a semicolon-separated
      list of parameter=value pairs.

5.3.  Offer/Answer Considerations

   The following considerations apply when using the SDP offer/answer
   RFC 3264 [RFC3264] mechanism to negotiate the "channels" attribute.

   o  If the offering endpoint specifies a value for the optional
      channels parameter greater than one and the answering endpoint
      both understands the parameter and cannot support that value
      requested, the answer MUST contain the optional channels parameter
      with the highest value it can support.

   o  If the offering endpoint specifies a value for the optional
      channels parameter the answer MUST contain the optional channels
      parameter unless the only value the answering endpoint can support
      is one, in which case the answer MAY contain the optional channels
      parameter with value of 1.

   o  If the offering endpoint specifies a value for the ptime parameter
      that the answering endpoint cannot support, the answer MUST
      contain the optional ptime parameter.

   o  If the offering endpoint specifies a value for the maxptime
      parameter that the answering endpoint cannot support, the answer
      MUST contain the optional maxptime parameter.





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5.4.  SDP Examples

   The following examples illustrate how to signal G.711.0 via SDP.

5.4.1.  SDP Example 1

         m=audio RTP/AVP 98
         a=rtpmap:98 G711-0/8000
         a=fmtp:98 complaw=mu

   In the above example the dynamic payload type 98 is mapped to G.711.0
   via the "a=rtpmap" parameter.  The mandatory "complaw" is on the
   "a=fmtp" parameter line.  Note that neither optional parameters
   "ptime" nor "channels" is present; although it is generally good form
   to include "ptime" in the SDP if the session is a constant ptime
   session for diagnostic purposes.

5.4.2.  SDP Example 2

   The following example illustrates an offering endpoint requesting 2
   channels, but the answering endpoint can only support (or render) one
   channel.

   Offer:

         m=audio RTP/AVP 98
         a=rtpmap:98 G711-0/8000/2
         a=ptime:20
         a=fmtp:98 complaw=al

   Answer:

         m=audio RTP/AVP 98
         a=rtpmap: 98 G711-0/8000/1
         a=ptime: 20
         a=fmtp:98 complaw=al

   In this example the offer had an optional channels parameter.  The
   answer must have the optional channels parameter also unless the
   value in the answer is one.  Shown here is when the answer explicitly
   contains the channels parameter (it need not have and it would be
   interpreted as one channel).  As mentioned previously, it is
   considered good form to include "ptime" in the SDP for session
   diagnostic purposes if the session is a constant ptime session.







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6.  G.711.0 Storage Mode Conventions and Definition

   The G.711.0 storage mode definition in this section is similar to
   many other IETF codecs (e.g., iLBC, EVRC-NW) and is essentially a
   concatenation of individual G.711.0 frames.

   We note that something must be stored for any G.711.0 frames that are
   not received at the receiving endpoint, no matter what the cause.  In
   this section we describe two mechanisms, a "G.711.0 PLC Frame" and a
   "G.711.0 Erasure Frame".  These G.711.0 PLC and G.711.0 Erasure
   Frames are described prior to the G.711.0 storage mode definition for
   clarity.

6.1.  G.711.0 PLC Frame

   When G.711 RTP payloads not received by a rendering endpoint a Packet
   Loss Concealment (PLC) mechanism is typically employed to "fill in"
   the missing G.711 symbols with something that is auditorially
   pleasing and thus the loss may be not noticed by a listener.  Such a
   PLC mechanism for G.711 is specified in ITU-T Rec. G.711 - Appendix 1
   [G.711-AP1].

   An natural extension when creating G.711.0 frames for storage
   environments is to employ such a PLC mechanism to create G.711
   symbols for the span of time in which G.711.0 payloads were not
   received - and then to compress the resulting "G.711 PLC symbols" via
   G.711.0 compression.  The G.711.0 frame(s) created by such a process
   are called "G.711.0 PLC Frames".

   Since PLC mechanisms are designed to render missing audio data with
   the best fidelity and intelligibility, G.711.0 frames created via
   such processing is likely best for most recording situations (such as
   voicemail storage) unless there is a requirement not to fabricate
   (audio) data not actually received.

   After such PLC G.711 symbols have been generated and then encoded by
   a G.711.0 encoder, the resulting frames may be stored in G.711.0
   frame format.  As a result, there is nothing to specify here - the
   G.711.0 PLC Frames are stored as if they were received by the
   receiving endpoint.  In other words, PLC-generated G.711.0 frames
   appear as "normal" or "ordinary" G.711.0 frames in the storage mode
   file.

6.2.  G.711.0 Erasure Frame

   "Erasure Frames", or equivalently "Null Frames", have been designed
   for many frame-based codecs since G.711 was standardized.  These
   null/erasure frames explicitly represent data from incoming audio



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   that were either not received by the receiving system or represent
   data that a transmitting system decided not to send.  Transmitting
   systems may choose not to send data for a variety of reasons (e.g.,
   not enough wireless link capacity in radio-based systems) and can
   choose to send a "null frame" in lieu of the actual audio.  It is
   also envisioned that erasure frames would be used in storage mode
   applications for specific archival purposes where there is a
   requirement not to fabricate audio data that was not actually
   received.

   Thus, a G.711.0 erasure frame is a representation of the amount of
   time in G.711.0 frames that were not received or not encoded by the
   transmitting system.

   Prior to defining a G.711.0 erasure frame it is beneficial to note
   what many G.711 RTP systems send when the endpoint is "muted".  When
   muted, many of these systems will send an entire G.711 payload of
   either 0+ or 0- (i.e., one of the two levels closest to "analog zero"
   in either G.711 companding law).  Next we note that a desirable
   property for a G.711.0 erasure frame is for "non G.711.0 Erasure
   Frame aware" endpoints to be able to playback a G.711.0 erasure frame
   with the existing G.711.0 ITU-T reference code.

   A G.711.0 Erasure Frame is defined as any G.711.0 frame for which the
   corresponding G.711 sample values are either the value 0++ or the
   value 0-- for the entirety of the G.711.0 frame.  The levels of 0++
   and 0-- are defined to be the two levels above or below analog zero,
   respectively.  An entire frame of value 0++ or 0-- is expected to be
   extraordinarily rare when the frame was in fact generated by a
   natural signal, as analog inputs such as speech and music are zero-
   mean and are typically acoustically coupled to digital sampling
   systems.  Note that the playback of a G.711.0 frame characterized as
   an erasure frame is auditorially equivalent to a muted signal (a very
   low value constant).

   These G.711.0 erasure frames can be reasonably characterized as null
   or erasure frames while meeting the desired playback goal of being
   decoded by the G.711.0 ITU-T reference code.  Thus, similarly to
   G.711 PLC frames, the G.711.0 erasure frames appear as "normal" or
   "ordinary" G.711.0 frames in the storage mode format.

6.3.  G.711.0 Storage Mode Definition

   The storage format is used for storing G.711.0 encoded frames.  The
   format for the G.711.0 storage mode file defined by this RFC is shown
   below.





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                        G.711.0 Storage Mode Format


          |---------------------------|----------|--------------|
          |       Magic Number        |          |              |
          |                           |  Version | Concatenated |
          | "#!G7110A\n" (for A-law)  |   Octet  |   G.711.0    |
          |            or             |          |    Frames    |
          | "#!G7110M\n" (for mu-law) |  "0x00"  |              |
          |___________________________|__________|______________|


                                 Figure 5

   The storage mode file consists of a magic number and a version octet
   followed by the individual G.711.0 frames concatenated together.

   The magic number for G.711.0 A-law corresponds to the ASCII character
   string "#!G7110A\n", i.e., "0x23 0x21 0x47 0x37 0x31 0x31 0x30 0x41
   0x0A".  Likewise, the magic number for G.711.0 MU-law corresponds to
   the ASCII character string "#!G7110M\n", i.e., "0x23 0x21 0x47 0x37
   0x31 0x31 0x4E 0x4D 0x0A".

   The version number octet allows for the future specification of other
   G.711.0 storage mode formats.  The specification of other storage
   mode formats may be desirable as G.711.0 frames are of variable
   length and a future format may include an indexing methodology that
   would enable playout far into a long G.711.0 recording without the
   necessity of decoding all the G.711.0 frames since the beginning of
   the recording.  Other future format specification may include support
   for multiple channels, metadata and the like.  For these reasons it
   was determined that a versioning strategy was desirable for the
   G.711.0 storage mode definition specified by this RFC.  This RFC only
   specifies Version 0 and thus the value of "0x00" MUST be used for the
   storage mode defined by this RFC.

   The G.711.0 codec data frames, including any necessary erasure or PLC
   frames, are stored in consecutive order concatenated together as
   shown in Section 4.2.2.  As the Version 0 storage mode only supports
   a single channel, the RTP payload format supporting multiple channels
   defined in Section 4.2.4 is not supported in this storage mode
   definition.

   To decode the individual G.711.0 frames, the algorithm presented in
   Section 4.2.2 may be used to decode the individual G.711.0 frames.
   If the version octet is determined not to be zero, the remainder of
   the payload MUST NOT be passed to the G.711.0 decoder, as the ITU-T
   G.711.0 reference decoder can only decode concatenated G.711.0 frames



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   and has not been designed to decode elements in yet to be specified
   future storage mode formats.

7.  Acknowledgements

   There have been many people contributing to G.711.0 in the course of
   its development.  The people listed here deserve special mention:
   Takehiro Moriya, Claude Lamblin, Herve Taddei, Simao Campos, Yusuke
   Hiwasaki, Jacek Stachurski, Lorin Netsch, Paul Coverdale, Patrick
   Luthi, Paul Barrett, Jari Hagqvist, Pengjun (Jeff) Huang, John Gibbs,
   Yutaka Kamamoto, and Csaba Kos.  The review and oversight by the IETF
   Payload Working Group chairs Ali Begen and Roni Even during the
   development of this RFC is appreciated.  Additionally, the careful
   review by Richard Barnes and extensive review by David Black and the
   rest of the IESG is likewise very much appreciated.

8.  Contributors

   The authors thank everyone who have contributed to this document.
   The people listed here deserve special mention: Ali Begen, Roni Even,
   and Hadriel Kaplan.

9.  IANA Considerations

   One media type (audio/G711-0) has been defined and requires IANA
   registration in the media types registry.  See Section 5.1 for
   details.

10.  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 in any appropriate RTP profile (for
   example RFC 3551 [RFC3551] or [RFC4585]).  This implies that
   confidentiality of the media streams is achieved by encryption; for
   example, through the application of SRTP [RFC3711].  Because the data
   compression used with this payload format is applied end-to-end, any
   encryption needs to be performed after compression.

   Note that the appropriate mechanism to ensure confidentiality and
   integrity of RTP packets and their payloads is very dependent on the
   application and on the transport and signaling protocols employed.
   Thus, although SRTP is given as an example above, other possible
   choices exist.

   Note that end-to-end security with either authentication, integrity
   or confidentiality protection will prevent a network element not
   within the security context from performing media-aware operations



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   other than discarding complete packets.  To allow any (media-aware)
   intermediate network element to perform its operations, it is
   required to be a trusted entity which is included in the security
   context establishment.

   G.711.0 has no known denial-of-service attacks due to decoding, as
   data posing as a desired G711.0 payload will be decoded into
   something (as per the decoding algorithm) with a finite amount of
   computation.  This is due to the decompression algorithm having a
   finite worst-case processing path (no infinite computational loops
   are possible).  We also note that the data read by the G.711.0
   decoder is controlled by the length of the individual encoded G.711.0
   frame(s) contained in the RTP payload.  The decoding algorithm
   specified in Section 4.2.3 above ensures that the G.711.0 decoder
   will not read beyond the length of the internal buffer specified
   (which is in turn specified to be no greater than the largest
   possible G.711.0 frame of 321 octets).  Therefore a G.711.0 payload
   does not carry "active content" that could impose malicious side-
   effects upon the receiver.

   G.711.0 is a variable bit rate (VBR) audio codec.  There have been
   recent concerns with VBR speech codecs where a passive observer can
   identify phrases from a standard speech corpus by means of the
   lengths produced by the encoder even when the payload is encrypted
   [IEEE].  In this paper, it was determined that some code excited
   linear prediction (CELP) codecs would produce discrete packet lengths
   for some phonemes.  And furthermore with the use of appropriately
   designed Hidden Markov Models (HMMs) that such a system could predict
   phrases with unexpected accuracy.  One CELP codec studied, SPEEX, had
   the property that it produced 21 different packet lengths in its
   wideband mode and that these packet lengths probabilistically mapped
   to phonemes that a HMM system could be trained on.  In this paper it
   was determined that a mitigation technique would be to pad the output
   of the encoder with random padding lengths to the effect: 1) that
   more discrete payload sizes would result, and 2) that the
   probabilistic mapping to phonemes would become less clear.  As G.711
   is not a speech model based codec, neither is G.711.0.  A G.711.0
   encoding, during talking periods, produces frames of varying frame
   lengths which are not likely to have a strong mapping to phonemes.
   Thus G.711.0 is not expected to have this same vulnerability.  It
   should be noted that "silence" (only one value of G.711 in the entire
   G.711 input frame)" or "near silence" (only a few G.711 values) is
   easily detectable as G.711.0 frame lengths or one or a few octets.
   If one desires to mitigate for silence/non-silence detection,
   statistically variable padding should be added to G.711.0 frames that
   resulted in very small G.711.0 frames (less than about 20% of the
   symbols of the corresponding G.711 input frame).  Methods of




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   introducing padding in the G.711.0 payloads have been provided in the
   G.711.0 RTP payload definition in Section 4.2.2.

11.  Congestion Control

   The G.711 codec is a Constant Bit Rate (CBR) codec which does not
   have a means to regulate the bitrate.  The G.711.0 lossless
   compression algorithm typically compresses the G.711 CBR stream into
   a lower bandwidth VBR stream.  However, being lossless, it does not
   possess means of further reducing the bitrate beyond the
   G.711.0-based compression result.  The G.711.0 RTP payloads can be
   made arbitrarily large by means of adding optional padding bytes
   (subject only to MTU limitations).

   Therefore, there are no explicit ways to regulate the bit-rate of the
   transmissions outlined in this RTP Payload format except by means of
   modulating the number of optional padding bytes in the RTP payload.

12.  References

12.1.  Normative References

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

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

   [RFC6838]  Freed, N., Klensin, J., and T. Hansen, "Media Type
              Specifications and Registration Procedures", BCP 13, RFC
              6838, January 2013.

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

   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
              Video Conferences with Minimal Control", STD 65, RFC 3551,
              July 2003.

   [RFC4585]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
              "Extended RTP Profile for Real-time Transport Control
              Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July
              2006.

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



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

   [G.711.0]  ITU-T G.711.0, , "Recommendation ITU-T G.711.0 - Lossless
              Compression of G.711 Pulse Code Modulation", September
              2009.

   [G.711]    ITU-T G.711.0, , "Recommendation ITU-T G.711: Pulse Code
              Modulation (PCM) of Voice Frequencies", November 1988.

   [G.711-AP1]
              ITU-T G.711 Appendix 1, , "Recommendation G.711
              Appendix 1: A high quality low-complexity algorithm for
              packet loss concealment with G.711", September 1999.

   [G.711-A1]
              ITU-T G.711 Amendment 1, , "Recommendation ITU-T G.711
              Amendment 1 - Amendment 1: New Annex A on Lossless
              Encoding of PCM Frames", September 2009.

12.2.  Informative References

   [G.729]    ITU-T G.729, , "Recommendation ITU-T G.729 - Coding of
              speech at 8 kbit/s using conjugate-structure algebraic-
              code-excited linear prediction (CS-ACELP)", January 2007.

   [G.722]    ITU-T G.722, , "Recommendation ITU-T G.722 - 7 kHz audio-
              coding within 64 kbit/s", November 1988.

   [ICASSP]   N. Harada, , Y. Yamamoto, , T. Moriya, , Y. Hiwasaki, , M.
              A. Ramalho, , L. Netsch, , Y. Stachurski, , Miao Lei, , H.
              Taddei, , and Q. Fengyan, "Emerging ITU-T Standard G.711.0
              - Lossless Compression of G.711 Pulse Code Modulation,
              International Conference on Acoustics Speech and Signal
              Processing (ICASSP), 2010, ISBN 978-1-4244-4244-4295-9",
              March 2010.

   [IEEE]     C.V. Wright, , L. Ballard, , S.E. Coull, , F. Monrose, ,
              and G.M. Masson, "Spot Me if You Can: Uncovering Spoken
              Phrases in Encrypted VoIP Conversations, IEEE Symposium on
              Security and Privacy, 2008, ISBN: 978-0-7695-3168-7", May
              2008.








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Authors' Addresses

   Michael A. Ramalho (editor)
   Cisco Systems, Inc.
   6310 Watercrest Way Unit 203
   Lakewood Ranch, FL  34202
   USA

   Phone: +1 919 476 2038
   Email: mramalho@cisco.com


   Paul E. Jones
   Cisco Systems, Inc.
   7025 Kit Creek Rd.
   Research Triangle Park, NC  27709
   USA

   Phone: +1 919 476 2048
   Email: paulej@packetizer.com


   Noboru Harada
   NTT Communications Science Labs.
   3-1 Morinosato-Wakamiya
   Atsugi, Kanagawa  243-0198
   JAPAN

   Phone: +81 46 240 3676
   Email: harada.noboru@lab.ntt.co.jp


   Muthu Arul Mozhi Perumal
   Ericsson
   Ferns Icon
   Doddanekundi, Mahadevapura
   Bangalore, Karnataka  560037
   India

   Phone: +91 9449288768
   Email: muthu.arul@gmail.com










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   Lei Miao
   Huawei Technologies Co. Ltd
   Q22-2-A15R, Enviroment Protection Park
   No. 156 Beiqing Road
   HaiDian District
   Beijing  100095
   China

   Phone: +86 1059728300
   Email: lei.miao@huawei.com









































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