Figure 6: G.729 Annex D bit packing
The net bit rate for the G.729 Annex E algorithm is 11.8 kbit/s and a
total of 118 bits are used. Two bits are appended as "don't care"
bits to complete an integer number of octets for the frame. For
G729E, the bits of a data frame are formatted as shown in the next
two diagrams (cf. Table E.1/G.729). The fields for the G729E forward
adaptive mode are packed as shown in Fig. 7.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0 0|L| L1 | L2 | L3 | P1 |P| C0_1|
| |0| | | | |0| |
| | |0 1 2 3 4 5 6|0 1 2 3 4|0 1 2 3 4|0 1 2 3 4 5 6 7| |0 1 2|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| | C1_1 | C2_1 | C3_1 | C4_1 |
| | | | | |
|3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| GA1 | GB1 | P2 | C0_2 | C1_2 | C2_2 |
| | | | | | |
|0 1 2|0 1 2 3|0 1 2 3 4|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| | C3_2 | C4_2 | GA2 | GB2 |DC |
| | | | | | |
|6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2|0 1 2 3|0 1|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 7: G.729 Annex E (forward adaptive mode) bit packing
The fields for the G729E backward adaptive mode are packed as shown
in Fig. 8.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1 1| P1 |P| C0_1 | C1_1 |
| | |0| 1 1 1| |
| |0 1 2 3 4 5 6 7|0|0 1 2 3 4 5 6 7 8 9 0 1 2|0 1 2 3 4 5 6 7|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| | C2_1 | C3_1 | C4_1 |GA1 | GB1 |P2 |
| | | | | | | |
|8 9|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2|0 1 2 3|0 1|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| | C0_2 | C1_2 | C2_2 |
| | 1 1 1| | |
|2 3 4|0 1 2 3 4 5 6 7 8 9 0 1 2|0 1 2 3 4 5 6 7 8 9|0 1 2 3 4 5|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| | C3_2 | C4_2 | GA2 | GB2 |DC |
| | | | | | |
|6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2|0 1 2 3|0 1|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 8: G.729 Annex E (backward adaptive mode) bit packing
4.5.8 GSM
GSM (Group Speciale Mobile) denotes the European GSM 06.10 standard
for full-rate speech transcoding, ETS 300 961, which is based on
RPE/LTP (residual pulse excitation/long term prediction) coding at a
rate of 13 kb/s [11,12,13]. The text of the standard can be obtained
from:
ETSI (European Telecommunications Standards Institute)
ETSI Secretariat: B.P.152
F-06561 Valbonne Cedex
France
Phone: +33 92 94 42 00
Fax: +33 93 65 47 16
Blocks of 160 audio samples are compressed into 33 octets, for an
effective data rate of 13,200 b/s.
4.5.8.1 General Packaging Issues
The GSM standard (ETS 300 961) specifies the bit stream produced by
the codec, but does not specify how these bits should be packed for
transmission. The packetization specified here has subsequently been
adopted in ETSI Technical Specification TS 101 318. Some software
implementations of the GSM codec use a different packing than that
specified here.
field field name bits field field name bits
________________________________________________
1 LARc[0] 6 39 xmc[22] 3
2 LARc[1] 6 40 xmc[23] 3
3 LARc[2] 5 41 xmc[24] 3
4 LARc[3] 5 42 xmc[25] 3
5 LARc[4] 4 43 Nc[2] 7
6 LARc[5] 4 44 bc[2] 2
7 LARc[6] 3 45 Mc[2] 2
8 LARc[7] 3 46 xmaxc[2] 6
9 Nc[0] 7 47 xmc[26] 3
10 bc[0] 2 48 xmc[27] 3
11 Mc[0] 2 49 xmc[28] 3
12 xmaxc[0] 6 50 xmc[29] 3
13 xmc[0] 3 51 xmc[30] 3
14 xmc[1] 3 52 xmc[31] 3
15 xmc[2] 3 53 xmc[32] 3
16 xmc[3] 3 54 xmc[33] 3
17 xmc[4] 3 55 xmc[34] 3
18 xmc[5] 3 56 xmc[35] 3
19 xmc[6] 3 57 xmc[36] 3
20 xmc[7] 3 58 xmc[37] 3
21 xmc[8] 3 59 xmc[38] 3
22 xmc[9] 3 60 Nc[3] 7
23 xmc[10] 3 61 bc[3] 2
24 xmc[11] 3 62 Mc[3] 2
25 xmc[12] 3 63 xmaxc[3] 6
26 Nc[1] 7 64 xmc[39] 3
27 bc[1] 2 65 xmc[40] 3
28 Mc[1] 2 66 xmc[41] 3
29 xmaxc[1] 6 67 xmc[42] 3
30 xmc[13] 3 68 xmc[43] 3
31 xmc[14] 3 69 xmc[44] 3
32 xmc[15] 3 70 xmc[45] 3
33 xmc[16] 3 71 xmc[46] 3
34 xmc[17] 3 72 xmc[47] 3
35 xmc[18] 3 73 xmc[48] 3
36 xmc[19] 3 74 xmc[49] 3
37 xmc[20] 3 75 xmc[50] 3
38 xmc[21] 3 76 xmc[51] 3
Table 2: Ordering of GSM variables
Octet Bit 0 Bit 1 Bit 2 Bit 3 Bit 4 Bit 5 Bit 6 Bit 7
_____________________________________________________________________
0 1 1 0 1 LARc0.0 LARc0.1 LARc0.2 LARc0.3
1 LARc0.4 LARc0.5 LARc1.0 LARc1.1 LARc1.2 LARc1.3 LARc1.4 LARc1.5
2 LARc2.0 LARc2.1 LARc2.2 LARc2.3 LARc2.4 LARc3.0 LARc3.1 LARc3.2
3 LARc3.3 LARc3.4 LARc4.0 LARc4.1 LARc4.2 LARc4.3 LARc5.0 LARc5.1
4 LARc5.2 LARc5.3 LARc6.0 LARc6.1 LARc6.2 LARc7.0 LARc7.1 LARc7.2
5 Nc0.0 Nc0.1 Nc0.2 Nc0.3 Nc0.4 Nc0.5 Nc0.6 bc0.0
6 bc0.1 Mc0.0 Mc0.1 xmaxc00 xmaxc01 xmaxc02 xmaxc03 xmaxc04
7 xmaxc05 xmc0.0 xmc0.1 xmc0.2 xmc1.0 xmc1.1 xmc1.2 xmc2.0
8 xmc2.1 xmc2.2 xmc3.0 xmc3.1 xmc3.2 xmc4.0 xmc4.1 xmc4.2
9 xmc5.0 xmc5.1 xmc5.2 xmc6.0 xmc6.1 xmc6.2 xmc7.0 xmc7.1
10 xmc7.2 xmc8.0 xmc8.1 xmc8.2 xmc9.0 xmc9.1 xmc9.2 xmc10.0
11 xmc10.1 xmc10.2 xmc11.0 xmc11.1 xmc11.2 xmc12.0 xmc12.1 xcm12.2
12 Nc1.0 Nc1.1 Nc1.2 Nc1.3 Nc1.4 Nc1.5 Nc1.6 bc1.0
13 bc1.1 Mc1.0 Mc1.1 xmaxc10 xmaxc11 xmaxc12 xmaxc13 xmaxc14
14 xmax15 xmc13.0 xmc13.1 xmc13.2 xmc14.0 xmc14.1 xmc14.2 xmc15.0
15 xmc15.1 xmc15.2 xmc16.0 xmc16.1 xmc16.2 xmc17.0 xmc17.1 xmc17.2
16 xmc18.0 xmc18.1 xmc18.2 xmc19.0 xmc19.1 xmc19.2 xmc20.0 xmc20.1
17 xmc20.2 xmc21.0 xmc21.1 xmc21.2 xmc22.0 xmc22.1 xmc22.2 xmc23.0
18 xmc23.1 xmc23.2 xmc24.0 xmc24.1 xmc24.2 xmc25.0 xmc25.1 xmc25.2
19 Nc2.0 Nc2.1 Nc2.2 Nc2.3 Nc2.4 Nc2.5 Nc2.6 bc2.0
20 bc2.1 Mc2.0 Mc2.1 xmaxc20 xmaxc21 xmaxc22 xmaxc23 xmaxc24
21 xmaxc25 xmc26.0 xmc26.1 xmc26.2 xmc27.0 xmc27.1 xmc27.2 xmc28.0
22 xmc28.1 xmc28.2 xmc29.0 xmc29.1 xmc29.2 xmc30.0 xmc30.1 xmc30.2
23 xmc31.0 xmc31.1 xmc31.2 xmc32.0 xmc32.1 xmc32.2 xmc33.0 xmc33.1
24 xmc33.2 xmc34.0 xmc34.1 xmc34.2 xmc35.0 xmc35.1 xmc35.2 xmc36.0
25 Xmc36.1 xmc36.2 xmc37.0 xmc37.1 xmc37.2 xmc38.0 xmc38.1 xmc38.2
26 Nc3.0 Nc3.1 Nc3.2 Nc3.3 Nc3.4 Nc3.5 Nc3.6 bc3.0
27 bc3.1 Mc3.0 Mc3.1 xmaxc30 xmaxc31 xmaxc32 xmaxc33 xmaxc34
28 xmaxc35 xmc39.0 xmc39.1 xmc39.2 xmc40.0 xmc40.1 xmc40.2 xmc41.0
29 xmc41.1 xmc41.2 xmc42.0 xmc42.1 xmc42.2 xmc43.0 xmc43.1 xmc43.2
30 xmc44.0 xmc44.1 xmc44.2 xmc45.0 xmc45.1 xmc45.2 xmc46.0 xmc46.1
31 xmc46.2 xmc47.0 xmc47.1 xmc47.2 xmc48.0 xmc48.1 xmc48.2 xmc49.0
32 xmc49.1 xmc49.2 xmc50.0 xmc50.1 xmc50.2 xmc51.0 xmc51.1 xmc51.2
Table 3: GSM payload format
In the GSM packing used by RTP, the bits SHALL be packed beginning
from the most significant bit. Every 160 sample GSM frame is coded
into one 33 octet (264 bit) buffer. Every such buffer begins with a
4 bit signature (0xD), followed by the MSB encoding of the fields of
the frame. The first octet thus contains 1101 in the 4 most
significant bits (0-3) and the 4 most significant bits of F1 (0-3) in
the 4 least significant bits (4-7). The second octet contains the 2
least significant bits of F1 in bits 0-1, and F2 in bits 2-7, and so
on. The order of the fields in the frame is described in Table 2.
4.5.8.2 GSM Variable Names and Numbers
In the RTP encoding we have the bit pattern described in Table 3,
where F.i signifies the ith bit of the field F, bit 0 is the most
significant bit, and the bits of every octet are numbered from 0 to 7
from most to least significant.
4.5.9 GSM-EFR
GSM-EFR denotes GSM 06.60 enhanced full rate speech transcoding,
specified in ETS 300 726 which is available from ETSI at the address
given in Section 4.5.8. This codec has a frame length of 244 bits.
For transmission in RTP, each codec frame is packed into a 31 octet
(248 bit) buffer beginning with a 4-bit signature 0xC in a manner
similar to that specified here for the original GSM 06.10 codec. The
packing is specified in ETSI Technical Specification TS 101 318.
4.5.10 L8
L8 denotes linear audio data samples, using 8-bits of precision with
an offset of 128, that is, the most negative signal is encoded as
zero.
4.5.11 L16
L16 denotes uncompressed audio data samples, using 16-bit signed
representation with 65,535 equally divided steps between minimum and
maximum signal level, ranging from -32,768 to 32,767. The value is
represented in two's complement notation and transmitted in network
byte order (most significant byte first).
The MIME registration for L16 in RFC3555 [7] specifies parameters
that MAY be used with MIME or SDP to indicate that analog pre-
emphasis was applied to the signal before quantization or to indicate
that a multiple-channel audio stream follows a different channel
ordering convention than is specified in Section 4.1.
4.5.12 LPC
LPC designates an experimental linear predictive encoding contributed
by Ron Frederick, which is based on an implementation written by Ron
Zuckerman posted to the Usenet group comp.dsp on June 26, 1992. The
codec generates 14 octets for every frame. The framesize is set to
20 ms, resulting in a bit rate of 5,600 b/s.
4.5.13 MPA
MPA denotes MPEG-1 or MPEG-2 audio encapsulated as elementary
streams. The encoding is defined in ISO standards ISO/IEC 11172-3
and 13818-3. The encapsulation is specified in RFC2250 [14].
The encoding may be at any of three levels of complexity, called
Layer I, II and III. The selected layer as well as the sampling rate
and channel count are indicated in the payload. The RTP timestamp
clock rate is always 90,000, independent of the sampling rate.
MPEG-1 audio supports sampling rates of 32, 44.1, and 48 kHz (ISO/IEC
11172-3, section 1.1; "Scope"). MPEG-2 supports sampling rates of
16, 22.05 and 24 kHz. The number of samples per frame is fixed, but
the frame size will vary with the sampling rate and bit rate.
The MIME registration for MPA in RFC3555 [7] specifies parameters
that MAY be used with MIME or SDP to restrict the selection of layer,
channel count, sampling rate, and bit rate.
4.5.14 PCMA and PCMU
PCMA and PCMU are specified in ITU-T Recommendation G.711. Audio
data is encoded as eight bits per sample, after logarithmic scaling.
PCMU denotes mu-law scaling, PCMA A-law scaling. A detailed
description is given by Jayant and Noll [15]. Each G.711 octet SHALL
be octet-aligned in an RTP packet. The sign bit of each G.711 octet
SHALL correspond to the most significant bit of the octet in the RTP
packet (i.e., assuming the G.711 samples are handled as octets on the
host machine, the sign bit SHALL be the most significant bit of the
octet as defined by the host machine format). The 56 kb/s and 48
kb/s modes of G.711 are not applicable to RTP, since PCMA and PCMU
MUST always be transmitted as 8-bit samples.
See Section 4.1 regarding silence suppression.
4.5.15 QCELP
The Electronic Industries Association (EIA) & Telecommunications
Industry Association (TIA) standard IS-733, "TR45: High Rate Speech
Service Option for Wideband Spread Spectrum Communications Systems",
defines the QCELP audio compression algorithm for use in wireless
CDMA applications. The QCELP CODEC compresses each 20 milliseconds
of 8,000 Hz, 16-bit sampled input speech into one of four different
size output frames: Rate 1 (266 bits), Rate 1/2 (124 bits), Rate 1/4
(54 bits) or Rate 1/8 (20 bits). For typical speech patterns, this
results in an average output of 6.8 kb/s for normal mode and 4.7 kb/s
for reduced rate mode. The packetization of the QCELP audio codec is
described in [16].
4.5.16 RED
The redundant audio payload format "RED" is specified by RFC2198
[17]. It defines a means by which multiple redundant copies of an
audio packet may be transmitted in a single RTP stream. Each packet
in such a stream contains, in addition to the audio data for that
packetization interval, a (more heavily compressed) copy of the data
from a previous packetization interval. This allows an approximation
of the data from lost packets to be recovered upon decoding of a
subsequent packet, giving much improved sound quality when compared
with silence substitution for lost packets.
4.5.17 VDVI
VDVI is a variable-rate version of DVI4, yielding speech bit rates of
between 10 and 25 kb/s. It is specified for single-channel operation
only. Samples are packed into octets starting at the most-
significant bit. The last octet is padded with 1 bits if the last
sample does not fill the last octet. This padding is distinct from
the valid codewords. The receiver needs to detect the padding
because there is no explicit count of samples in the packet.
It uses the following encoding:
DVI4 codeword VDVI bit pattern
_______________________________
0 00
1 010
2 1100
3 11100
4 111100
5 1111100
6 11111100
7 11111110
8 10
9 011
10 1101
11 11101
12 111101
13 1111101
14 11111101
15 11111111
5. Video
The following sections describe the video encodings that are defined
in this memo and give their abbreviated names used for
identification. These video encodings and their payload types are
listed in Table 5.
All of these video encodings use an RTP timestamp frequency of 90,000
Hz, the same as the MPEG presentation time stamp frequency. This
frequency yields exact integer timestamp increments for the typical
24 (HDTV), 25 (PAL), and 29.97 (NTSC) and 30 Hz (HDTV) frame rates
and 50, 59.94 and 60 Hz field rates. While 90 kHz is the RECOMMENDED
rate for future video encodings used within this profile, other rates
MAY be used. However, it is not sufficient to use the video frame
rate (typically between 15 and 30 Hz) because that does not provide
adequate resolution for typical synchronization requirements when
calculating the RTP timestamp corresponding to the NTP timestamp in
an RTCP SR packet. The timestamp resolution MUST also be sufficient
for the jitter estimate contained in the receiver reports.
For most of these video encodings, the RTP timestamp encodes the
sampling instant of the video image contained in the RTP data packet.
If a video image occupies more than one packet, the timestamp is the
same on all of those packets. Packets from different video images
are distinguished by their different timestamps.
Most of these video encodings also specify that the marker bit of the
RTP header SHOULD be set to one in the last packet of a video frame
and otherwise set to zero. Thus, it is not necessary to wait for a
following packet with a different timestamp to detect that a new
frame should be displayed.
5.1 CelB
The CELL-B encoding is a proprietary encoding proposed by Sun
Microsystems. The byte stream format is described in RFC2029 [18].
5.2 JPEG
The encoding is specified in ISO Standards 10918-1 and 10918-2. The
RTP payload format is as specified in RFC2435 [19].
5.3 H261
The encoding is specified in ITU-T Recommendation H.261, "Video codec
for audiovisual services at p x 64 kbit/s". The packetization and
RTP-specific properties are described in RFC2032 [20].
5.4 H263
The encoding is specified in the 1996 version of ITU-T Recommendation
H.263, "Video coding for low bit rate communication". The
packetization and RTP-specific properties are described in RFC2190
[21]. The H263-1998 payload format is RECOMMENDED over this one for
use by new implementations.
5.5 H263-1998
The encoding is specified in the 1998 version of ITU-T Recommendation
H.263, "Video coding for low bit rate communication". The
packetization and RTP-specific properties are described in RFC2429
[22]. Because the 1998 version of H.263 is a superset of the 1996
syntax, this payload format can also be used with the 1996 version of
H.263, and is RECOMMENDED for this use by new implementations. This
payload format does not replace RFC2190, which continues to be used
by existing implementations, and may be required for backward
compatibility in new implementations. Implementations using the new
features of the 1998 version of H.263 MUST use the payload format
described in RFC2429.
5.6 MPV
MPV designates the use of MPEG-1 and MPEG-2 video encoding elementary
streams as specified in ISO Standards ISO/IEC 11172 and 13818-2,
respectively. The RTP payload format is as specified in RFC2250
[14], Section 3.
The MIME registration for MPV in RFC3555 [7] specifies a parameter
that MAY be used with MIME or SDP to restrict the selection of the
type of MPEG video.
5.7 MP2T
MP2T designates the use of MPEG-2 transport streams, for either audio
or video. The RTP payload format is described in RFC2250 [14],
Section 2.
5.8 nv
The encoding is implemented in the program `nv', version 4, developed
at Xerox PARC by Ron Frederick. Further information is available
from the author:
Ron Frederick
Blue Coat Systems Inc.
650 Almanor Avenue
Sunnyvale, CA 94085
United States
EMail: ronf@bluecoat.com
6. Payload Type Definitions
Tables 4 and 5 define this profile's static payload type values for
the PT field of the RTP data header. In addition, payload type
values in the range 96-127 MAY be defined dynamically through a
conference control protocol, which is beyond the scope of this
document. For example, a session directory could specify that for a
given session, payload type 96 indicates PCMU encoding, 8,000 Hz
sampling rate, 2 channels. Entries in Tables 4 and 5 with payload
type "dyn" have no static payload type assigned and are only used
with a dynamic payload type. Payload type 2 was assigned to G721 in
RFC1890 and to its equivalent successor G726-32 in draft versions of
this specification, but its use is now deprecated and that static
payload type is marked reserved due to conflicting use for the
payload formats G726-32 and AAL2-G726-32 (see Section 4.5.4).
Payload type 13 indicates the Comfort Noise (CN) payload format
specified in RFC3389 [9]. Payload type 19 is marked "reserved"
because some draft versions of this specification assigned that
number to an earlier version of the comfort noise payload format.
The payload type range 72-76 is marked "reserved" so that RTCP and
RTP packets can be reliably distinguished (see Section "Summary of
Protocol Constants" of the RTP protocol specification).
The payload types currently defined in this profile are assigned to
exactly one of three categories or media types: audio only, video
only and those combining audio and video. The media types are marked
in Tables 4 and 5 as "A", "V" and "AV", respectively. Payload types
of different media types SHALL NOT be interleaved or multiplexed
within a single RTP session, but multiple RTP sessions MAY be used in
parallel to send multiple media types. An RTP source MAY change
payload types within the same media type during a session. See the
section "Multiplexing RTP Sessions" of RFC3550 for additional
explanation.
PT encoding media type clock rate channels
name (Hz)
___________________________________________________
0 PCMU A 8,000 1
1 reserved A
2 reserved A
3 GSM A 8,000 1
4 G723 A 8,000 1
5 DVI4 A 8,000 1
6 DVI4 A 16,000 1
7 LPC A 8,000 1
8 PCMA A 8,000 1
9 G722 A 8,000 1
10 L16 A 44,100 2
11 L16 A 44,100 1
12 QCELP A 8,000 1
13 CN A 8,000 1
14 MPA A 90,000 (see text)
15 G728 A 8,000 1
16 DVI4 A 11,025 1
17 DVI4 A 22,050 1
18 G729 A 8,000 1
19 reserved A
20 unassigned A
21 unassigned A
22 unassigned A
23 unassigned A
dyn G726-40 A 8,000 1
dyn G726-32 A 8,000 1
dyn G726-24 A 8,000 1
dyn G726-16 A 8,000 1
dyn G729D A 8,000 1
dyn G729E A 8,000 1
dyn GSM-EFR A 8,000 1
dyn L8 A var. var.
dyn RED A (see text)
dyn VDVI A var. 1
Table 4: Payload types (PT) for audio encodings
PT encoding media type clock rate
name (Hz)
_____________________________________________
24 unassigned V
25 CelB V 90,000
26 JPEG V 90,000
27 unassigned V
28 nv V 90,000
29 unassigned V
30 unassigned V
31 H261 V 90,000
32 MPV V 90,000
33 MP2T AV 90,000
34 H263 V 90,000
35-71 unassigned ?
72-76 reserved N/A N/A
77-95 unassigned ?
96-127 dynamic ?
dyn H263-1998 V 90,000
Table 5: Payload types (PT) for video and combined
encodings
Session participants agree through mechanisms beyond the scope of
this specification on the set of payload types allowed in a given
session. This set MAY, for example, be defined by the capabilities
of the applications used, negotiated by a conference control protocol
or established by agreement between the human participants.
Audio applications operating under this profile SHOULD, at a minimum,
be able to send and/or receive payload types 0 (PCMU) and 5 (DVI4).
This allows interoperability without format negotiation and ensures
successful negotiation with a conference control protocol.
7. RTP over TCP and Similar Byte Stream Protocols
Under special circumstances, it may be necessary to carry RTP in
protocols offering a byte stream abstraction, such as TCP, possibly
multiplexed with other data. The application MUST define its own
method of delineating RTP and RTCP packets (RTSP [23] provides an
example of such an encapsulation specification).
8. Port Assignment
As specified in the RTP protocol definition, RTP data SHOULD be
carried on an even UDP port number and the corresponding RTCP packets
SHOULD be carried on the next higher (odd) port number.
Applications operating under this profile MAY use any such UDP port
pair. For example, the port pair MAY be allocated randomly by a
session management program. A single fixed port number pair cannot
be required because multiple applications using this profile are
likely to run on the same host, and there are some operating systems
that do not allow multiple processes to use the same UDP port with
different multicast addresses.
However, port numbers 5004 and 5005 have been registered for use with
this profile for those applications that choose to use them as the
default pair. Applications that operate under multiple profiles MAY
use this port pair as an indication to select this profile if they
are not subject to the constraint of the previous paragraph.
Applications need not have a default and MAY require that the port
pair be explicitly specified. The particular port numbers were
chosen to lie in the range above 5000 to accommodate port number
allocation practice within some versions of the Unix operating
system, where port numbers below 1024 can only be used by privileged
processes and port numbers between 1024 and 5000 are automatically
assigned by the operating system.
9. Changes from RFC1890
This RFCrevises RFC1890. It is mostly backwards-compatible with
RFC1890 except for functions removed because two interoperable
implementations were not found. The additions to RFC1890 codify
existing practice in the use of payload formats under this profile.
Since this profile may be used without using any of the payload
formats listed here, the addition of new payload formats in this
revision does not affect backwards compatibility. The changes are
listed below, categorized into functional and non-functional changes.
Functional changes:
o Section 11, "IANA Considerations" was added to specify the
registration of the name for this profile. That appendix also
references a new Section 3 "Registering Additional Encodings"
which establishes a policy that no additional registration of
static payload types for this profile will be made beyond those
added in this revision and included in Tables 4 and 5. Instead,
additional encoding names may be registered as MIME subtypes for
binding to dynamic payload types. Non-normative references were
added to RFC3555 [7] where MIME subtypes for all the listed
payload formats are registered, some with optional parameters for
use of the payload formats.
o Static payload types 4, 16, 17 and 34 were added to incorporate
IANA registrations made since the publication of RFC1890, along
with the corresponding payload format descriptions for G723 and
H263.
o Following working group discussion, static payload types 12 and 18
were added along with the corresponding payload format
descriptions for QCELP and G729. Static payload type 13 was
assigned to the Comfort Noise (CN) payload format defined in RFC
3389. Payload type 19 was marked reserved because it had been
temporarily allocated to an earlier version of Comfort Noise
present in some draft revisions of this document.
o The payload format for G721 was renamed to G726-32 following the
ITU-T renumbering, and the payload format description for G726 was
expanded to include the -16, -24 and -40 data rates. Because of
confusion regarding draft revisions of this document, some
implementations of these G726 payload formats packed samples into
octets starting with the most significant bit rather than the
least significant bit as specified here. To partially resolve
this incompatibility, new payload formats named AAL2-G726-16, -24,
-32 and -40 will be specified in a separate document (see note in
Section 4.5.4), and use of static payload type 2 is deprecated as
explained in Section 6.
o Payload formats G729D and G729E were added following the ITU-T
addition of Annexes D and E to Recommendation G.729. Listings
were added for payload formats GSM-EFR, RED, and H263-1998
published in other documents subsequent to RFC1890. These
additional payload formats are referenced only by dynamic payload
type numbers.
o The descriptions of the payload formats for G722, G728, GSM, VDVI
were expanded.
o The payload format for 1016 audio was removed and its static
payload type assignment 1 was marked "reserved" because two
interoperable implementations were not found.
o Requirements for congestion control were added in Section 2.
o This profile follows the suggestion in the revised RTP spec that
RTCP bandwidth may be specified separately from the session
bandwidth and separately for active senders and passive receivers.
o The mapping of a user pass-phrase string into an encryption key
was deleted from Section 2 because two interoperable
implementations were not found.
o The "quadrophonic" sample ordering convention for four-channel
audio was removed to eliminate an ambiguity as noted in Section
4.1.
Non-functional changes:
o In Section 4.1, it is now explicitly stated that silence
suppression is allowed for all audio payload formats. (This has
always been the case and derives from a fundamental aspect of
RTP's design and the motivations for packet audio, but was not
explicit stated before.) The use of comfort noise is also
explained.
o In Section 4.1, the requirement level for setting of the marker
bit on the first packet after silence for audio was changed from
"is" to "SHOULD be", and clarified that the marker bit is set only
when packets are intentionally not sent.
o Similarly, text was added to specify that the marker bit SHOULD be
set to one on the last packet of a video frame, and that video
frames are distinguished by their timestamps.
o RFCreferences are added for payload formats published after RFC
1890.
o The security considerations and full copyright sections were
added.
o According to Peter Hoddie of Apple, only pre-1994 Macintosh used
the 22254.54 rate and none the 11127.27 rate, so the latter was
dropped from the discussion of suggested sampling frequencies.
o Table 1 was corrected to move some values from the "ms/packet"
column to the "default ms/packet" column where they belonged.
o Since the Interactive Multimedia Association ceased operations, an
alternate resource was provided for a referenced IMA document.
o A note has been added for G722 to clarify a discrepancy between
the actual sampling rate and the RTP timestamp clock rate.
o Small clarifications of the text have been made in several places,
some in response to questions from readers. In particular:
- A definition for "media type" is given in Section 1.1 to allow
the explanation of multiplexing RTP sessions in Section 6 to be
more clear regarding the multiplexing of multiple media.
- The explanation of how to determine the number of audio frames
in a packet from the length was expanded.
- More description of the allocation of bandwidth to SDES items
is given.
- A note was added that the convention for the order of channels
specified in Section 4.1 may be overridden by a particular
encoding or payload format specification.
- The terms MUST, SHOULD, MAY, etc. are used as defined in RFC
2119.
o A second author for this document was added.
10. Security Considerations
Implementations using the profile defined in this specification are
subject to the security considerations discussed in the RTP
specification [1]. This profile does not specify any different
security services. The primary function of this profile is to list a
set of data compression encodings for audio and video media.
Confidentiality of the media streams is achieved by encryption.
Because the data compression used with the payload formats described
in this profile is applied end-to-end, encryption may be performed
after compression so there is no conflict between the two operations.
A potential denial-of-service threat exists for data encodings using
compression techniques that have non-uniform receiver-end
computational load. The attacker can inject pathological datagrams
into the stream which are complex to decode and cause the receiver to
be overloaded.
As with any IP-based protocol, in some circumstances a receiver may
be overloaded simply by the receipt of too many packets, either
desired or undesired. Network-layer authentication MAY be used to
discard packets from undesired sources, but the processing cost of
the authentication itself may be too high. In a multicast
environment, source pruning is implemented in IGMPv3 (RFC3376) [24]
and in multicast routing protocols to allow a receiver to select
which sources are allowed to reach it.
11. IANA Considerations
The RTP specification establishes a registry of profile names for use
by higher-level control protocols, such as the Session Description
Protocol (SDP), RFC2327 [6], to refer to transport methods. This
profile registers the name "RTP/AVP".
Section 3 establishes the policy that no additional registration of
static RTP payload types for this profile will be made beyond those
added in this document revision and included in Tables 4 and 5. IANA
may reference that section in declining to accept any additional
registration requests. In Tables 4 and 5, note that types 1 and 2
have been marked reserved and the set of "dyn" payload types included
has been updated. These changes are explained in Sections 6 and 9.
12. References
12.1 Normative References
[1] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson,
"RTP: A Transport Protocol for Real-Time Applications", RFC
3550, July 2003.
[2] Bradner, S., "Key Words for Use in RFCs to Indicate Requirement
Levels", BCP 14, RFC2119, March 1997.
[3] Apple Computer, "Audio Interchange File Format AIFF-C", August
1991. (also ftp://ftp.sgi.com/sgi/aiff-c.9.26.91.ps.Z).
12.2 Informative References
[4] Braden, R., Clark, D. and S. Shenker, "Integrated Services in
the Internet Architecture: an Overview", RFC1633, June 1994.
[5] Blake, S., Black, D., Carlson, M., Davies, E., Wang, Z. and W.
Weiss, "An Architecture for Differentiated Service", RFC2475,
December 1998.
[6] Handley, M. and V. Jacobson, "SDP: Session Description
Protocol", RFC2327, April 1998.
[7] Casner, S. and P. Hoschka, "MIME Type Registration of RTP
Payload Types", RFC3555, July 2003.
[8] Freed, N., Klensin, J. and J. Postel, "Multipurpose Internet
Mail Extensions (MIME) Part Four: Registration Procedures", BCP
13, RFC2048, November 1996.
[9] Zopf, R., "Real-time Transport Protocol (RTP) Payload for
Comfort Noise (CN)", RFC3389, September 2002.
[10] Deleam, D. and J.-P. Petit, "Real-time implementations of the
recent ITU-T low bit rate speech coders on the TI TMS320C54X
DSP: results, methodology, and applications", in Proc. of
International Conference on Signal Processing, Technology, and
Applications (ICSPAT) , (Boston, Massachusetts), pp. 1656--1660,
October 1996.
[11] Mouly, M. and M.-B. Pautet, The GSM system for mobile
communications Lassay-les-Chateaux, France: Europe Media
Duplication, 1993.
[12] Degener, J., "Digital Speech Compression", Dr. Dobb's Journal,
December 1994.
[13] Redl, S., Weber, M. and M. Oliphant, An Introduction to GSM
Boston: Artech House, 1995.
[14] Hoffman, D., Fernando, G., Goyal, V. and M. Civanlar, "RTP
Payload Format for MPEG1/MPEG2 Video", RFC2250, January 1998.
[15] Jayant, N. and P. Noll, Digital Coding of Waveforms--Principles
and Applications to Speech and Video Englewood Cliffs, New
Jersey: Prentice-Hall, 1984.
[16] McKay, K., "RTP Payload Format for PureVoice(tm) Audio", RFC
2658, August 1999.
[17] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., Handley, M.,
Bolot, J.-C., Vega-Garcia, A. and S. Fosse-Parisis, "RTP Payload
for Redundant Audio Data", RFC2198, September 1997.
[18] Speer, M. and D. Hoffman, "RTP Payload Format of Sun's CellB
Video Encoding", RFC2029, October 1996.
[19] Berc, L., Fenner, W., Frederick, R., McCanne, S. and P. Stewart,
"RTP Payload Format for JPEG-Compressed Video", RFC2435,
October 1998.
[20] Turletti, T. and C. Huitema, "RTP Payload Format for H.261 Video
Streams", RFC2032, October 1996.
[21] Zhu, C., "RTP Payload Format for H.263 Video Streams", RFC2190,
September 1997.
[22] Bormann, C., Cline, L., Deisher, G., Gardos, T., Maciocco, C.,
Newell, D., Ott, J., Sullivan, G., Wenger, S. and C. Zhu, "RTP
Payload Format for the 1998 Version of ITU-T Rec. H.263 Video
(H.263+)", RFC2429, October 1998.
[23] Schulzrinne, H., Rao, A. and R. Lanphier, "Real Time Streaming
Protocol (RTSP)", RFC2326, April 1998.
[24] Cain, B., Deering, S., Kouvelas, I., Fenner, B. and A.
Thyagarajan, "Internet Group Management Protocol, Version 3",
RFC3376, October 2002.
13. Current Locations of Related Resources
Note: Several sections below refer to the ITU-T Software Tool
Library (STL). It is available from the ITU Sales Service, Place des
Nations, CH-1211 Geneve 20, Switzerland (also check
http://www.itu.int). The ITU-T STL is covered by a license defined
in ITU-T Recommendation G.191, "Software tools for speech and audio
coding standardization".
DVI4
An archived copy of the document IMA Recommended Practices for
Enhancing Digital Audio Compatibility in Multimedia Systems (version
3.0), which describes the IMA ADPCM algorithm, is available at:
http://www.cs.columbia.edu/~hgs/audio/dvi/
An implementation is available from Jack Jansen at
ftp://ftp.cwi.nl/local/pub/audio/adpcm.shar
G722
An implementation of the G.722 algorithm is available as part of the
ITU-T STL, described above.
G723
The reference C code implementation defining the G.723.1 algorithm
and its Annexes A, B, and C are available as an integral part of
Recommendation G.723.1 from the ITU Sales Service, address listed
above. Both the algorithm and C code are covered by a specific
license. The ITU-T Secretariat should be contacted to obtain such
licensing information.
G726
G726 is specified in the ITU-T Recommendation G.726, "40, 32, 24, and
16 kb/s Adaptive Differential Pulse Code Modulation (ADPCM)". An
implementation of the G.726 algorithm is available as part of the
ITU-T STL, described above.
G729
The reference C code implementation defining the G.729 algorithm and
its Annexes A through I are available as an integral part of
Recommendation G.729 from the ITU Sales Service, listed above. Annex
I contains the integrated C source code for all G.729 operating
modes. The G.729 algorithm and associated C code are covered by a
specific license. The contact information for obtaining the license
is available from the ITU-T Secretariat.
GSM
A reference implementation was written by Carsten Bormann and Jutta
Degener (then at TU Berlin, Germany). It is available at
http://www.dmn.tzi.org/software/gsm/
Although the RPE-LTP algorithm is not an ITU-T standard, there is a C
code implementation of the RPE-LTP algorithm available as part of the
ITU-T STL. The STL implementation is an adaptation of the TU Berlin
version.
LPC
An implementation is available at
ftp://parcftp.xerox.com/pub/net-research/lpc.tar.Z
PCMU, PCMA
An implementation of these algorithms is available as part of the
ITU-T STL, described above.
14. Acknowledgments
The comments and careful review of Simao Campos, Richard Cox and AVT
Working Group participants are gratefully acknowledged. The GSM
description was adopted from the IMTC Voice over IP Forum Service
Interoperability Implementation Agreement (January 1997). Fred Burg
and Terry Lyons helped with the G.729 description.
15. Intellectual Property Rights Statement
The IETF takes no position regarding the validity or scope of any
intellectual property or other rights that might be claimed to
pertain to the implementation or use of the technology described in
this document or the extent to which any license under such rights
might or might not be available; neither does it represent that it
has made any effort to identify any such rights. Information on the
IETF's procedures with respect to rights in standards-track and
standards-related documentation can be found in BCP-11. Copies of
claims of rights made available for publication and any assurances of
licenses to be made available, or the result of an attempt made to
obtain a general license or permission for the use of such
proprietary rights by implementors or users of this specification can
be obtained from the IETF Secretariat.
The IETF invites any interested party to bring to its attention any
copyrights, patents or patent applications, or other proprietary
rights which may cover technology that may be required to practice
this standard. Please address the information to the IETF Executive
Director.
16. Authors' Addresses
Henning Schulzrinne
Department of Computer Science
Columbia University
1214 Amsterdam Avenue
New York, NY 10027
United States
EMail: schulzrinne@cs.columbia.edu
Stephen L. Casner
Packet Design
3400 Hillview Avenue, Building 3
Palo Alto, CA 94304
United States
EMail: casner@acm.org
17. Full Copyright Statement
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