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RFC 7655

 
 
 

RTP Payload Format for G.711.0

Part 2 of 2, p. 10 to 32
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4.  RTP Header and Payload

   In this section, we describe the precise format for G.711.0 frames
   carried via RTP.  We begin with an 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 that 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).

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   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
   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 (e.g., see [RFC3550] and
      [RFC4585]).

      SN - As per [RFC3550]

      timestamp - As per [RFC3550]

      SSRC - As per [RFC3550]

      CSRC - As per [RFC3550]

   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 are as typically used with
   G.711.  PT (payload type) is as defined in [RFC3551].

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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 in which
   carrying only one G.711.0 frame in the RTP payload is desired.  This
   case is expected to be the dominant use case and is shown separately
   for the purposes of clarity.

   The second example, in Section 4.2.2, depicts the general case in
   which carrying one or more G.711.0 frames in the RTP payload is
   desired.  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.

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

            Figure 2: Single G.711.0 Frame in RTP Payload Case

   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 the Security Considerations (see
   Section 8).

   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

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   the G.711.0 decoding process.  The decoding process is fully
   described in Section 4.2.3.

4.2.2.  G.711.0 RTP Payload Definition

   This section defines the G.711.0 RTP payload and illustrates the case
   in which 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, this case is
   identical to the previous example.

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

         Figure 3: One or More G.711.0 Frames in RTP Payload Case

   We note here that when we have multiple G.711.0 frames, 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 8).  Additional details
   about the encoding process 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 a frequency of 8000 samples 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 octets 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 that usually results in the highest compression) or
   the combination of possible input frames to a G.711.0 encoder that
   results 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, 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 is
         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 octets 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 that 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 something 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 Steps H3 and H4 as separate steps (i.e., Step H5
   immediately follows H2) at the added computational cost of some

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   additional buffer passing to/from the G.711.0 frame decoder
   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 frame boundary encoded by G.711.0.  Thus, Steps H3
   and 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.

   Section 4 of RFC 3551 [RFC3551] provides guidelines for encoding
   audio channels and Section 4.1 of RFC 3551 [RFC3551] for the ordering
   of the channels within the RTP payload.  The ordering guidelines in
   Section 4.1 of RFC 3551 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 an 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).

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   Section 4.2 of RFC 3551 provides guidelines for sample-based
   encodings such as G.711.  This guidance is tantamount to interleaving
   the 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 stated
   assumption that "the frame size for the frame-oriented codecs is
   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 of this
   document) 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 8).  This is
   depicted in Figure 4.

           |----------|---------|----------|---------|---------|
           | 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: Multiple G.711.0 Channel Superframes in RTP Payload

   We note that although the individual superframes can be of different
   lengths in octets (and usually are), the number of G.711 source
   symbols represented -- in compressed form -- in each channel
   superframe is identical (since all the channels represent the
   identically same time interval).

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   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, the number of channels of data
   contained in the payload could be determined.  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, 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
   SDP RFC 4566 [RFC4566] is also provided for those applications that
   use SDP.

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5.1.  Media Type Registration

   Type name: audio

   Subtype name: G711-0

   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
      Section 4.1 of RFC 3551 [RFC3551] 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.

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   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 8.

   Interoperability considerations: none

   Published specification:

      ITU-T Rec. G.711.0 and RFC 7655 (this document).

   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

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   Change controller:

      IETF Payload working group delegated from the IESG.

5.2.  Mapping to SDP Parameters

   The information carried in the media type specification has a
   specific mapping to fields in SDP, which is commonly used to describe
   an 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
   mechanism [RFC3264] to negotiate the "channels" attribute.

   o  If the offering endpoint specifies a value for the optional
      channels parameter that is 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 a 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.

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   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.

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 RFC 3951 [RFC3951] and EVRC-NW RFC
   6884 [RFC6884]), 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 are not received by a rendering endpoint, a
   PLC mechanism is typically employed to "fill in" the missing G.711
   symbols with something that is auditorially pleasing; 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].

   A 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.

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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
   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.

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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.

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

                   Figure 5: G.711.0 Storage Mode Format

   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

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

7.  IANA Considerations

   One media type (audio/G711-0) has been defined and registered in
   IANA's "Media Types" registry.  See Section 5.1 for details.

8.  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 applicable RTP profile (such as
   RTP/AVP [RFC3551], RTP/AVPF [RFC4585], RTP/SAVP [RFC3711], or RTP/
   SAVPF [RFC5124].  However, as "Securing the RTP Protocol Framework:
   Why RTP Does Not Mandate a Single Media Security Solution" [RFC7202]
   discusses, it is not a responsibility of the RTP payload format to
   discuss or mandate what solutions are used to meet the basic security
   goals like confidentiality, integrity, and source authenticity for
   RTP in general.  This responsibility lays on anyone using RTP in an
   application.  They can find guidance on available security mechanisms
   and important considerations in "Options for Securing RTP Sessions"
   [RFC7201].  Applications SHOULD use one or more appropriate strong
   security mechanisms.  The rest of this Security Considerations
   section discusses the security impacting properties of the playload
   format itself.

   Because the data compression used with this payload format is applied
   end-to-end, any encryption needs to be performed after compression.

   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
   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 that is included in the security
   context establishment.

   G.711.0 has no known denial-of-service (DoS) 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

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   specified previously in Section 4.2.3 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 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.
   Furthermore, with the use of appropriately designed Hidden Markov
   Models (HMMs), such a system could predict phrases with unexpected
   accuracy.  One CELP codec studied, SPEEX, had the property that
   produced 21 different packet lengths in its wideband mode, and these
   packet lengths probabilistically mapped to phonemes that an 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 that 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 introducing padding in the G.711.0 payloads have
   been provided in the G.711.0 RTP payload definition in Section 4.2.2.

9.  Congestion Control

   The G.711 codec is a Constant Bit Rate (CBR) codec that 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 compression result
   based on G.711.0.  The G.711.0 RTP payloads can be made arbitrarily
   large by means of adding optional padding bytes (subject only to MTU
   limitations).

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   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.

10.  References

10.1.  Normative References

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

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

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

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

   [RFC2119]   Bradner, S., "Key words for use in RFCs to Indicate
               Requirement Levels", BCP 14, RFC 2119,
               DOI 10.17487/RFC2119, March 1997,
               <http://www.rfc-editor.org/info/rfc2119>.

   [RFC3264]   Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
               with Session Description Protocol (SDP)", RFC 3264,
               DOI 10.17487/RFC3264, June 2002,
               <http://www.rfc-editor.org/info/rfc3264>.

   [RFC3550]   Schulzrinne, H., Casner, S., Frederick, R., and V.
               Jacobson, "RTP: A Transport Protocol for Real-Time
               Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
               July 2003, <http://www.rfc-editor.org/info/rfc3550>.

   [RFC3551]    Schulzrinne, H. and S. Casner, "RTP Profile for Audio
               and Video Conferences with Minimal Control", STD 65,
               RFC 3551, DOI 10.17487/RFC3551, July 2003,
               <http://www.rfc-editor.org/info/rfc3551>.

   [RFC3711]   Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
               Norrman, "The Secure Real-time Transport Protocol
               (SRTP)", RFC 3711, DOI 10.17487/RFC3711, March 2004,
               <http://www.rfc-editor.org/info/rfc3711>.

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   [RFC3951]   Andersen, S., Duric, A., Astrom, H., Hagen, R., Kleijn,
               W., and J. Linden, "Internet Low Bit Rate Codec (iLBC)",
               RFC 3951, DOI 10.17487/RFC3951, December 2004,
               <http://www.rfc-editor.org/info/rfc3951>.

   [RFC4566]   Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
               Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
               July 2006, <http://www.rfc-editor.org/info/rfc4566>.

   [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, DOI 10.17487/RFC4585, July 2006,
               <http://www.rfc-editor.org/info/rfc4585>.

   [RFC5124]   Ott, J. and E. Carrara, "Extended Secure RTP Profile for
               Real-time Transport Control Protocol (RTCP)-Based
               Feedback (RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124,
               February 2008, <http://www.rfc-editor.org/info/rfc5124>.

   [RFC6838]   Freed, N., Klensin, J., and T. Hansen, "Media Type
               Specifications and Registration Procedures", BCP 13,
               RFC 6838, DOI 10.17487/RFC6838, January 2013,
               <http://www.rfc-editor.org/info/rfc6838>.

   [RFC6884]   Fang, Z., "RTP Payload Format for the Enhanced Variable
               Rate Narrowband-Wideband Codec (EVRC-NW)", RFC 6884,
               DOI 10.17487/RFC6884, March 2013,
               <http://www.rfc-editor.org/info/rfc6884>.

   [RFC7201]   Westerlund, M. and C. Perkins, "Options for Securing RTP
               Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014,
               <http://www.rfc-editor.org/info/rfc7201>.

   [RFC7202]   Perkins, C. and M. Westerlund, "Securing the RTP
               Framework: Why RTP Does Not Mandate a Single Media
               Security Solution", RFC 7202, DOI 10.17487/RFC7202, April
               2014, <http://www.rfc-editor.org/info/rfc7202>.

10.2.  Informative References

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

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

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   [ICASSP]    Harada, N., Yamamoto, Y., Moriya, T., Hiwasaki, Y.,
               Ramalho, M., Netsch, L., Stachurski, J., Miao, L.,
               Taddei, H., and F. Qi, "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]      Wright, C., Ballard, L., Coull, S., Monrose, F., and G.
               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.

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, the extensive review by David Black, and
   the reviews provided by the IESG are likewise very much appreciated.

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.

Authors' Addresses

   Michael A. Ramalho (editor)
   Cisco Systems, Inc.
   6310 Watercrest Way Unit 203
   Lakewood Ranch, FL  34202
   United States
   Phone: +1 919 476 2038
   Email: mramalho@cisco.com

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   Paul E. Jones
   Cisco Systems, Inc.
   7025 Kit Creek Road
   Research Triangle Park, NC  27709
   United States

   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


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

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