RFC 7655
RTP Payload Format for G.711.0
Pages: 32
Proposed Standard
Proposed Standard
Part 2 of 2 – Pages 10 to 32
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).
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].
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
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.
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).
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.
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
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).
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).
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.
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.
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
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.
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.
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.
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.
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
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
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).
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>.
[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.
[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
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